Path: news1.ucsd.edu!ihnp4.ucsd.edu!swrinde!newsfeed.internetmci.com!in1.uu.net!usc!sdd.hp.com!hplabs!unix.sri.com!news.Stanford.EDU!newshub.internex.net!news.internex.net!tang.bdti.com!kabir
From: phil@bdti.com
Newsgroups: comp.dsp,comp.answers,news.answers
Subject: comp.dsp FAQ [1 of 3]
Supersedes: <compdsp.1_818463006@bdti.com>
Followup-To: poster
Date: 20 Dec 1995 00:59:32 GMT
Organization: Berkeley Design Technology, Inc.
Lines: 1570
Approved: news-answers-request@MIT.Edu
Distribution: world
Expires: 2 Feb 1996 01:04:40 GMT
Message-ID: <compdsp.1_819421480@bdti.com>
NNTP-Posting-Host: bdti.com
Summary: This is a periodic posting to comp.dsp that gives information
on frequently asked questions asked in this newsgroup.
Originator: kabir@tang.bdti.com
Xref: news1.ucsd.edu comp.dsp:20123 comp.answers:13192 news.answers:51481
Archive-name: dsp-faq/part1
Last-modified: Wed June 21 1995
Version: 2.2
FAQs (Frequently asked questions with answers) on Digital Signal
Processing
Version date: June 21, 1995
_________________________________________________________________
Note: This FAQ is based on material aged one to two years. We have
made efforts to correct some glaring errors, and have reformatted the
FAQ a bit. More to come. We would appreciate any suggestions or
observations. Contact us via e-mail at comp-dsp-faq@bdti.com.
- Phil, FAQ maintainers
_________________________________________________________________
0. What is comp.dsp?
1. General DSP.
1.1 DSP book and article references.
1.2 Where can I get free software for general DSP?
2. Algorithms and standards.
2.1 Where can I get some algorithms for DSP?
2.2 What are CELP and LPC? Where can I get source for them?
2.3 What is ADPCM? Where can I get source for it?
2.4 What is GSM? Where can I get source for it?
2.5 How does pitch perception work, and how do I implement it?
2.6 What standards exist for digital audio?
What is AES/EBU? What is S/PDIF?
2.7 What is mu-law encoding? Where can I get source for it?
2.8 How can I do CD DAT sample rate conversion?
3. Programmable DSP chips and their software.
3.1 What are some current, popular programmable DSP chips?
3.2 Software for Motorola DSPs.
3.3 Software for Texas Instruments DSPs.
3.4 Software for Analog Devices DSPs.
4. Hardware.
4.1 DSP development boards.
4.2 Who makes AES/EBU chips?
5. Operating Systems.
6. List of manufacturers, addresses, and telephone numbers.
7. Summary of FTP sites.
8. Wavelets Information.
People involved...
_________________________________________________________________
Q0: What is comp.dsp?
_________________________________________________________________
Comp.dsp is a worldwide UseNet news group that is used to discuss
various aspects of digital signal processing. It is unmoderated,
though we try to keep the signal to noise ratio up :-). If you need to
ask a question that isn't in the FAQ, and can't figure out how to
post, consult news.newusers.questions. Other relevant news groups are
comp.compression, comp.speech, and sci.image.processing.
_________________________________________________________________
Click on dsp_faq.zip or dsp_faq.tar.Z to download a compressed HTML
version of the FAQ.
Click on dsp_faq.asc.zip or dsp_faq.asc.tar.Z to download a compressed
ASCII version of the FAQ.
(When you click on these links, your browser should tell you that it
can't display the files and then ask you if you want to download them
instead. Say "yes." We're still setting up an anonymous FTP site. This
is just an interim solution. -- Phil)
It is also available on World Wide Web, which offers a much nicer
interface. Try: http://www.bdti.com/dsp_faq.htm
Q1.1: Summary of DSP books and significant research articles.
Q1.1.1: BIBLES OF DSP.
A.V. Oppenheim and R.W. Schafer, "Digital Signal
Processing", Prentice-Hall, Inc., Englewood Cliffs, N.J.,
1975.
A.V. Oppenheim and R.W. Schafer, "Discrete-Time Signal
Processing" Prentice Hall, Englewood Cliffs, New Jersey 07632,
1989. ISBN 0-13-216292-X This is an updated version of the
original, with some old material deleted and lots of new
material added.
L.R. Rabiner & R.W. Schafer, "Digital Processing of Speech
Signals", Prentice Hall, 1978, ISBN 0-13-213603-1.
R. E. Crochiere & L. R. Rabiner, "Multirate Digital Signal
Processing", Prentice-Hall, 1983, ISBN 0136051626. This book
is the only real reference for filter banks and multirate
systems, as opposed to being a tutorial.
P. P. Vaidyanathan, "Multirate Systems and Filter Banks",
Prentice-Hall. 911 pp.
Thomas Parsons, "Voice and Speech Processing", McGraw-Hill,
1987, ISBN 0-07-048541-0. Addresses the cocktail party
effect, as well as other material. [Maurice Givens,
maury@tellabs.com]
__________________________________________________________________________
Q1.1.2: Adaptive signal processing.
S. Haykin, "Adaptive Filter Theory", 2nd Ed., Prentice
Hall, Englewood Cliffs, NJ, 1991.
B. Widrow and S.D. Stearns, "Adaptive Signal Processing",
Prentice-Hall, Inc., Englewood Cliffs, N.J., 1985.
__________________________________________________________________________
Q1.1.3: Array signal processing.
J.E. Hudson, "Adaptive Array Principles", IEE London and
New York, Peter Peregrinus Ltd. Stevenage, U.K., and New York,
1981.
R.A. Monzingo and T.W. Miller, "Introduction to Adaptive
Arrays" John Wiley and Sons, New York, 1980.
S. Haykin, J.H. Justice, N.L. Owsley, J.L. Yen, and A.C. Kak
"Array Signal Processing", Prentice-Hall, Inc., Englewood
Cliffs, N.J., 1985.
D. H. Johnson and D. E. Dudgeon, Array Signal Processing, Concepts and
Techniques, Prentice-Hall, 1993
R.T. Compton, Jr., "Adaptive Antennas, Concepts and
Performance", Prentice-Hall, 1988, ISBN 0-13-004151-3.
__________________________________________________________________________
Q1.1.4: Windowing articles.
F. J. Harris, "On the Use of Windows for Harmonic
Analysis with the DFT", IEEE Proceedings, January 1978,
pp. 51-83. Perhaps the classic overview paper for
discrete-time windows. It discusses some 15 different classes
of windows including their spectral responses and the reasons
for their development. [Brian Evans, evans@eedsp.gatech.edu]
There are several typos in this paper. The errors are
corrected by A. H. Nuttall in "Some Windows with Very Good
Sidelobe Behavior," IEEE Trans. on Acoustics, Speech, and
Signal Processing, Vol. ASSP-29, No. 1, February 1981.
Nezih C. Geckinli & Davras Yavuz, "Some Novel Windows and a
Concise Tutorial Comparison of Window Families", IEEE
Transactions on Acoustics, Speech, and Signal Processing,
Vol. ASSP-26, No. 6, December 1978. [Bob Beauchaine,
bobb@vice.ico.tek.com]
Lineu C. Barbosa, "A Maximum-Energy-Concentration Spectral
Window," IBM J. Res. Develop., Vol. 30, No. 3, May 1986,
p. 321-325. An elegant method for designing a time-discrete
solution for realization of a spectral window which is ideal
from an energy concentration viewpoint. This window is one
that concentrates the maximum amount of energy in a specified
bandwidth and hence provides optimal spectral resolution.
Unlike the Kaiser window, this window is a discrete-time
realization having the same objectives as the continuous-time
prolate spheroidal function; at the expense of not having a
closed form solution. [Joe Campbell,
jpcampb@afterlife.ncsc.mil]
__________________________________________________________________________
Q1.1.5: Digital Audio Effects Processing.
Books (in no particular order, sorry):
Hal Chamberlin, Musical Applications of Microprocessors,
2nd Ed., Hayden Book Company, 1985.
Barry Blesser and J. Kates. "Digital Processing in Audio
Signals." In A. V. Oppenheim, ed. Applications of Digital
Signal Processing. Englewood Cliffs, NJ: Prentice-Hall, 1978.
Digital Signal Processing Committee of IEEE Acoustics, Speech,
and Signal Processing Society, ed. Programs for Digital Signal
Processing. New York: IEEE Press, 1979.
John Strawn, ed., "Digital Audio Signal Processing: An
Anthology.", Los Altos, CA: W. Kaufmann, 1985. [Contains
Moorer J.A. "About This Reverb..." and contains an article
which gives a code for Phase Vocoder -- great tool for EQ, for
Pitchshifter and more --Juhana Kouhia]
Charles Dodge and Thomas A. Jerse. Computer Music: Synthesis,
Composition, and Performance. New York: Schirmer Books, 1985.
F. Richard Moore, "Elements of Computer Music", Englewood
Cliffs, NJ: Prentice-Hall, 1990. ISBN: 0-13252-552-6
[Recommended. --Juhana Kouhia]
Curtis Roads and John Strawn, ed., "The Foundations of
Computer Music", Cambridge, MA: MIT Press, 1985. [Contains
article on analysis/synthesis by Strawn, recommended; also an
another article maybe by J.A. Moorer -- Juhana Kouhia]
John Strawn, ed., "Digital Audio Signal Processing", 283
pages, $34.95, ISBN 0-86576-082-9, pub: A-R Editions.
Contents:
1. Introduction to the mathematics of DSP (F. Richard Moore)
[Not a bad little text]
2. Introduction to digital filter theory (Julius O. Smith)
[Not a bad little text, either]
3. Spiral Synthesis (Tracy Lind Petersen)
[first published account of a new synthesis technique]
4. Signal processing aspects of computer music (J. A. Moorer)
[James Moorer's classic article--discusses many synthesis
techniques. Reverb algorithms. More than 6 pages of refs]
5. An introduction to the phase vocoder (J. W. Gordon, J. Strawn)
[Includes source code for a phase vocoder--a powerful method for
synthesis, pitch shifting, time scale modification, etc.]
[Comments by Quinn Jensen].
Curtis Road, ed., "Composers and the Computer", 201 pages,
$27.95, ISBN 0-86576-085-3, pub: A-R Editions.
John Strawn, ed., "Digital Audio Engineering", 144 pages,
$29.95, ISBN 0-86576-087-X pub: A-R Editions.
Deta S. Davis, "Computer Applications in Music: A
Bibliography", 537 pages, $49.95, ISBN 0-89579-225-7, pub: A-R
Editions.
Ken C. Pohlmann, "The Compact Disc: A Handbook of Theory and
Use", 288 pages, $45.95 (cloth) ISBN 0-89579-234-6, $29.95
(paper) ISBN 0-89579-228-1, pub: A-R Editions.
Forthcoming books:
Curtis Roads, "A Computer Music History: Musical Automation
from Antiquity to the Computer Age"
Joseph Rothstein, "MIDI: A Comprehensive Introduction"
David Cope, "Computer Analysis of Musical Style"
Dexter Morrill and Rick Taube, "A Little Book of Computer
Music Instruments"
Articles:
James A. Moorer, "About This Reverberation Business", Computer
Music Journal 3, 20 (1979): 13-28. (Also in Foundations of CM
below). [Ok article, but you have to know basic DSP
operations. --Juhana Kouhia]
Check more articles from Journal of the Audio Engineering
Society (JAES), for example more articles by Strawn.
Note: books published by A-R editions can be ordered from:
A-R Editions
801 Deming Way
Madison, Wisconsin 53717
608-836-9000 (They accept VISA orders)
[The above is largely from Quinn Jensen, jensenq@qcj.icon.com;
Juhana Kouhia, jk87377@cc.tut.fi; William Alves,
alves@calvin.usc.edu; and Paul A Simoneau,
pas1@kepler.unh.edu]
Q1.2: Where can I get free software for general DSP?
The packages listed below are mostly for general purpose DSP, that is,
DSP that is not specific to a particular programmable DSP chip. See
the later sections in the FAQ for software relevant to a particular
programmable DSP chip.
According to Brian Evans: "There was an entire session on this subject
[free DSP software] at ICASSP '92, chaired by Dr. Sally Wood and Dr.
James McClellan. It appears in Volume 4 of the Proceedings, pages
73-112. There was another such session at ICASSP '93." [Brian Evans,
evans@eedsp.gatech.edu] Much of the information below is from Brian's
mail.
_________________________________________________________________
Q1.2.1: What is Gabriel? Where can I get it?
Gabriel was a hierarchical block diagram environment for prototyping
signal processing systems on single or multiple processors. It has
been superceded by Ptolemy (see below), and is no longer available.
_________________________________________________________________
Q1.2.2: What is Ptolemy? Where can I get it?
Description: Ptolemy provides a highly flexible foundation
for the specification, simulation, and rapid prototyping of
systems. It is an object oriented framework within which
diverse models of computation can co-exist and interact. For
example, using Ptolemy a data-flow system can be easily
connected to a hardware simulator which in turn may be
connected to a discrete-event system, etc. Because of this,
Ptolemy can be used to model entire systems.
In addition, Ptolemy now has code generation capabilities.
From a flow graph description, Ptolemy can generate both C
code and DSP assembly code for rapid prototyping. Note that
code generation is not yet complete, and is included in the
current release for demonstration purposes only.
Ptolemy has been used for a broad range of applications
including signal processing, telecomunications, parallel
processing, wireless communications, network design, radio
astronomy, real-time systems, and hardware/software
co-design. Ptolemy has also been used as a lab for signal
processing and communications courses. Currently Ptolemy has
hundreds of users in over 100 sites, both in industry and
academia.
Ptolemy is available for the Sun 4 (sparc), DecStation
(MIPS), and HP (HP-PA) architectures. Installing the system
requires 90 Mbytes for Ptolemy (more if you optionally
remake). Ptolemy also requires at least 8 Mbytes of physical
memory. Linux binaries also exist.
Ptolemy is available via anonymous ftp. Get the file:
file://ptolemy.eecs.berkeley.edu/pub/README
and follow the instructions.
ptolemy.eecs.berkeley.edu contains the entire
Ptolemy distribution, a postscript version of the Ptolemy
manual, and several Ptolemy papers.
Organizations without Internet FTP capability can obtain
Ptolemy, without support, from ILP. This is often a more stable,
less featured version than is available by FTP.
EECS/ERL Industrial Liaison Program Office
Software Distribution
205 Cory Hall
University of California, Berkeley
Berkeley, CA 94720
(510) 643-6687
email: ilpsoftware@eecs.berkeley.edu
This includes printed documentation, including installation
instructions, a user's guide, and manual pages. A handling
fee (on the order of $250) will be charged.
Contact: Alan Kamas, aok@ohm.berkeley.edu.
_________________________________________________________________
Q1.2.3: What is Khoros? Where can I get it?
Description: Block diagram simulator for image and video
processing. 1-D signal processing is also supported. See
the UseNet group comp.soft-sys.khoros.
Platforms: sun 3, sun 4, others? X windows. Written in C.
To obtain, get this file, and read the instructions:
file://ftp.eece.unm.edu/pub/khoros/release/install.ftp
_________________________________________________________________
Q1.2.4: What are DSP Tutorials? Where can I get them?
Package: DSP Tutorials
Description: Computer aided instruction.
Platforms: suns under SunView.
Contact: Dr. Sally Wood, Electrical Engineering Department,
Santa Clara University, Santa Clara, CA 95053.
_________________________________________________________________
Q1.2.5: What are some DSP extensions to MATLAB? Where can
I get them? Package: MATLAB user's group public domain
extensions to MATLAB
Description: The MATLAB Digest is issued at irregular
intervals based on the number of questions and software items
contributed by users. To make submissions to the digest,
please send to hwilson@ua1vm.ua.edu with a subject: "DIG" and
description.
For the Pacific, try netlib@draci.cs.uow.edu.au located at
the University of Wollongong, NSW, Australia.
A plethora of toolboxes are available at FTP site:
research.att.com (use netlib for the username)
_________________________________________________________________
General index for the MATLAB User Group software library
Currently there are the following subdirectories:
approximation approximation theory
archive old MATLAB user group digests
control control theory
dataanalysis data analysis and statistics
graphics graphics programs
integration numerical integration
linearalgebra linear algebra utilities
misc miscellaneous
ode ordinary differential equations
optimization as the name says
pde partial differential equations
rootfinding zero-finding routines
specialfunctions special functions
teaching for classroom use
tools miscellaneous tools
In order to get an index for a subdirectory (tools, say) send
the message:
send index from MATLAB/tools
to netlib@ornl.gov.
In order to get some code, (unbundle in the `tools'
directory, say), send the message:
send unbundle from MATLAB/tools
to netlib@ornl.gov.
_________________________________________________________________
There is a set of Wavelet Tools available for MATLAB, see
Section 8 of this FAQ.
_________________________________________________________________
Communications Toolbox
We have developed a "Communications Toolbox" based on the
Matlab code for classroom use. It is used by students taking
a 4th year communications course where the emphasis is on
digital coding of waveforms and on digital data transmission
systems. The Matlab code that constitutes this toolbox has
been in use for over two years.
There are close to 100 "M-files" that implement various
functions. Some of them are quite simple and are based on
existing Matlab M-files. But a great many of them has been
created from scratch. We also prepared a lab manual (in TEX
format) for the 7 simulations which the students perform as
the lab component of this course. The topics of these
simulations are:
[1]. Probability Theory
[2]. Random Processes
[3]. Quantization
[4]. Binary Signalling Formats
[5]. Detection
[6]. Digital Modulation
[7]. Digital Communication
New version (Matlab 4.1) is available on:
file://ftp.mathworks.com/pub/contrib/misc/comm_tbx.tar
Old version (Matlab 3.5) is available on:
file://evans.ee.adfa.oz.au/pub/matlab/comms/comm_tbx.tar
Functionality is basically the same.
The manual has also been slightly changed. I am still
working to get all the figures in postscript format. Please
continue using the old manual until I have the new manual in
postscript format ready.
[Mehmet Zeytinoglu - mzeytin@ee.ryerson.ca]
_________________________________________________________________
FOR STUDENTS: Prentice Hall has published a student edition
of matlab which contains a book and set of disks for PCs and
Macs. The software is limited only in matrix size (32 x 32
matrix; 1024 elements) and in its ability to import or call C
or Fortran subroutines. On the plus side, it is able to run
without a coprocessor (it will use one if it is present) and
it includes a subset of the Signal Processing and Controls
Toolboxes, The Signals and Systems Toolbox, which provides
for added functionality.
Book only (about US$30): ISBN =0138560064;
Book + disk: (about US$50) ISBN=0-13-855974-0 for 3.5"
or ISBN=0-13-855982-1 for 5.25
Macintosh version: ISBN=0-13-855990-2.
There will be related books out by mid to late 1993 :
Computer Aided Signal Processing with MATLAB, by Burrus,
Oppenheim, McClellan, Parks, Schafer, and Schussler;
and Signal Processing : A Computer Approach, by Etter. More
books in this MATLAB Curriculum Series are planned.
For general info: matlab@prenhall.com
[From the Matlab Users Group (Editor, hwilson@ua1vm.ua.edu)]
_________________________________________________________________
Q1.2.6: What are the Signal Processing Packages for Mathematica?
Where can I get them?
Package: Signal Processing Packages (SPP) and Notebooks.
Freely distributable extensions to Mathematica. Enables the
symbolic manipulation of signal processing expressions: 1-D
discrete/continuous convolutions and 1-D/m-D linear transforms
(Laplace, Fourier, z, DTFT, and DFT). For linear transforms,
you can specify your own transform pairs and see the
intermediate computations. Great for showing students how to
take transforms, or for deriving input-output relationships in
a transform domain. Additional abilities include analog
filter design, solving DE's using transforms, converting
signal processing expressions to their equivalent TeX forms,
number theoretic operations (Bezout numbers, Smith Form
decompositions, and matrix factors), and multirate operations
(graphical design of 2-d decimators). Accompanying the SPPs
are tutorial notebooks on analog filter design, Fourier
analysis, piecewise convolution, and the z-transform (includes
a discussion of fundamentals of digital filter design). These
Notebooks illustrate difficult concepts (such as the
flip-and-slide view of convolution) through animation.
Get this file, and read the instructions:
file://gauss.eedsp.gatech.edu/Mathematica/README
A freely distributable Notebook reader is available for
Macintosh computers and IBM-compatibles running MicroSoft
Windows by anonymous ftp:
Mac: file://mathsource.wri.com/pub/NumberedItems/0204-297-0011
Windows: file://mathsource.wri.com/pub/NumberedItems/0203-599-0011
Contact: Brian Evans, evans@eedsp.gatech.edu.
_________________________________________________________________
Dr. Roberto H. Bamberger reports: I have developed a series
of about 30 Lectures that I use for EE341 (Analog
Communication Systems) here at Washington State University.
They use the SPP by Brian Evans. They discuss many concepts
associated with linear systems theory. They are available
from: file://yardbird.eecs.wsu.edu/pub/Notebooks. Topics
covered include LTI system theory, convolution, AM, FM, PM
modulation and demodulation, and the sampling theorem. NOTE:
All Notebooks were developed under NeXTSTEP 3.1 using
Mathematica 2.2. I make no guarantees about the graphics
being able to be rendered on anything other than a NeXT.
_________________________________________________________________
FOR STUDENTS: A student version of Mathematica is available
for $175. The price includes a copy of the reference manual.
The only drawbacks to the student version are that the
floating point coprocessor is disabled and that upgrades
cannot be ordered.
_________________________________________________________________
Q1.2.7: What is the Control Systems Analysis Packages for Mathematica?
Where can I get them?
Package: Control Systems Analysis Package (COSYPAK) and Notebooks
Description: Public domain extension to Mathematica.
Classical and state-space control analysis and design
methods. The Notebooks supplement the material in the
textbook "Modern Controls Theory" by Ogata. Largely based on
the Signal Processing Packages (SPP, see above).
Contact: Dr. Sreenath, sree@veda.esys.cwru.edu.
To obtain: anonymous ftp veda.esys.cwru.edu (129.22.40.9).
_________________________________________________________________
Q1.2.8: What are some other Mathematica DSP Notebooks?
The following Mathematica notebooks can be ftped from
ftp.apple.com:
pub/malcolm/FilterDesign.math IIR Filter Design (continuous and discrete)
pub/malcolm/ear.math Implementation of Lyon's Cochlear Model
pub/malcolm/Gammatone.math Implementation of Gammatone Cochlear Model
Printed copies (with floppies) are available from the author
[malcolm@apple.com]
The following Mathematica notebooks can be ftped from
ccrma-ftp.stanford.edu:
pub/DSP/GenHamming.ma.Z Generalized Hamming windows
pub/DSP/Kaiser.ma.Z The Kaiser window
pub/DSP/WinFlt.ma.Z Digital filter design by the "window method"
(There are other DSP related items in pub/DSP on ccrma-ftp;
see other sections of this FAQ for details).
_________________________________________________________________
Q1.2.9: What is the Linear Systems Toolbox for Maple?
Where can I get it?
Package: Linear systems toolbox for Maple.
Description: Public domain extension to Maple.
Contact: Tony Richardson, amr@mpl.ucsd.edu.
To obtain:
file://ftp.egr.duke.edu/pub/maple/linsys1.2.tar.Z
_________________________________________________________________
Q1.2.10: Where can I get text to speech conversion software?
Free (but not public domain) text to speech conversion
software is available via anonymous ftp from
wilma.cs.brown.edu in the pub directory as speak.tar.Z. It
will compile and run on a SPARC's built-in audio after
modifying speak.c with the path of your libaudio.h (e.g.,
/usr/demo/SOUND/libaudio.h). It's a simple phoneme
concatenation system with commensurate synthesized speech
quality (a directory of phoneme audio files is included).
[Joe Campbell, jpcampb@afterlife.ncsc.mil]
A public domain version of the same Naval Research Lab text
to phoneme rules can be obtained from:
file://svr-ftp.eng.cam.ac.uk/comp.speech/sources/english2phoneme.shar
A implementation of the Klatt phoneme to waveform speech
synthesiser is in:
file://svr-ftp.eng.cam.ac.uk/comp.speech/sources/klatt-0.02.tar.Z
This directory also contains lossless speech compression
(shorten-1.08.tar.Z), speech recognition (recnet-1.1.tar),
acoustic modelling (rasta.tar.Z) and text normalisation
(textnorm.shar) software.
_________________________________________________________________
Q1.2.11: Where can I get filter design software? There
are filter design programs available via anonymous FTP. The
following are summarized here and discussed in greater detail
below:
1. August 1992 IEEE Trans. on Signal Processing: METEOR FIR
filter design program.
2. DFIR FIR filter design program.
3. Netlib IIR filter design.
4. IEEE Press "Programs for Digital Signal Processing".
_________________________________________________________________
The August 92 issue of IEEE Transactions on Signal Processing
there is a paper entitled "METEOR: A Constraint-Based FIR
Filter Design Program" by Kenneth Steiglitz, Thomas W. Parks
and James F. Kaiser. They describe an FIR design program
which allows specification of the target frequency response
characteristics in a fairly generalised and flexible way. As
well as designing filters, the program can optimise filter
lengths and push band limits.
The paper contains a footnote which says "Pascal and C
versions of source code are available to anonymous ftp at
princeton.edu in the directory /pub as meteor.p, form.p,
meteor.c and form.c".
True, they are. They appear to work. The Pascal versions
have been put through p2c to get the C versions; all the
needed Pascal library stuff is included in the C versions and
they built error-free out of the box for me on an SGI
machine.
One catch is, there is no manual - you need the paper to know
how to drive the programs.
[Steve Clift, clift@ml.csiro.au]
_________________________________________________________________
Another public domain filter design package is DFIR, for FIR
filter designs. It includes design capabilities for:
equiripple linear phase multiband filters, linear phase
differentiators, linear phase Hilbert transform filters, MMSE
interpolating filters and equiripple Nyquist filters. It is
written in Fortran 77 and has been tested on DECStations and
Suns.
It is available from: aldebaran.ee.mcgill.ca/pub/dfir.
Additionally, a package to plot filter responses is available
in "pltfilter-V2R0.tar.Z".
[Peter Kabal, via Witold Waldman]
_________________________________________________________________
Another source is netlib: "A free program to design IIR
Butterworth, Chebyshev, and Cauer (elliptic) filters, in any
of lowpass, bandpass, band reject, and high pass
configurations, is available in netlib (e.g.
research.att.com) as the file netlib/cephes/ellf.shar.Z. By
email to netlib@research.att.com the request message text is
`send ellf from cephes'
[Stephen Moshier, mosher@world.std.com]
_________________________________________________________________
The Fortran source code from the IEEE Press book "Programs
For Digital Signal Processing" is available for anonymous ftp
from
file://soma.crl.mcmaster.ca/pub/IEEE/software/dsp.zip
or
file://soma.crl.mcmaster.ca/pub/IEEE/software/dsp.tar.gz
It includes FIR and IIR filter design software, FFT
subroutines, interpolation programs, a coherence and
cross-spectral estimation program, linear prediction analysis
programs, and a frequency domain filtering program.
There is also a C/C++ version of the Parks-McLellan FIR
filter design program available from
file://ftp.uu.net/usenet/comp.sources.misc/volume22/fir/part01.Z
This program was created and tested using Borland C++ 2.0.
This requires a pretty reasonable C++ compiler - it is
reported that QuickC (not C++) won't do it.
[Witold Waldman, witold@hotblk.aed.dsto.gov.au, from Charles
Owen at mgcbo@uxa.ecn.bgu.au]
[also Andrew Ukrainec, andy@array.ca]
_________________________________________________________________
{ There are other free filter design programs floating around
out there, such as optfir/wfir. Does anyone know of ftp sites? }
_________________________________________________________________
Q1.2.12 What is PC Convolution? Where can I get it?
P.C. convolution is a educational software package that graphically
demonstrates the convolution operation. It runs on IBM PC type computers
using DOS 4.0 or later. It is currently being used in schools of Mathematics,
Electrical Engineering, Earth Sciences, Aeronautics, Astronomy, Geophysics,
and (believe it or not) Experimental Psychology.
The current version of this software only demonstrates continuous time
convolution, but a discrete time version is in the works.
Anyone may download a demonstration version of this software via anonymous
ftp from 131.151.4.11 (file name /pub/pc_conv.zip)
University instructors my obtain a free, fully operational version by
contacting Dr. Kurt Kosbar at the address listed below.
Dr. Kurt Kosbar
117 Electrical Engineering Building, University of Missouri - Rolla
Rolla, Missouri, USA 65401, phone: (314) 341-4894
e-mail: kk@ee.umr.edu
_________________________________________________________________
Q1.2.13: What is the AudioFile System? Where can I get
it?
The AudioFile System (AF) is a device-independent
network-transparent audio server. The distribution includes
device drivers and server code for Digital RISC systems
running Ultrix, Digital Alpha AXP systems running OSF/1, and
Sun Microsystems SPARCstations running SunOS. Also included
are an API and library, out-of-the-box core applications, and
a number of contributed applications. AudioFile allows
applications to generate and process audio in real-time and
at present handles up to 48 KHz stereo audio.
AudioFile is distributed in source form, with a copyright
allowing unrestricted use for any purpose except sale (see
the Copyright notice). af@crl.dec.com is a mailing list for
discussions of AudioFile. Send mail to af-request@crl.dec.com
to be added to this list.
The kit is located at:
file://crl.dec.com/pub/DEC/AF/AF2R2.tar.Z
A sample kit of sound-bites is available as:
file://crl.dec.com/pub/DEC/AF/AF2R2-other.tar.Z
[Larry Stewart, stewart@crl.dec.com]
_________________________________________________________________
1.2.14 What is MathViews? Where can I get it?
Package-Name: mathview.zip
MathViews for Windows/32 - Math Software for Windows
(32-bit). Current version is 1.60. "MathViews for
Windows/32 is Matlab look-alike. It has a full set of linear
algebra and signal processing functionality."
No sources. Windows 3.1. Shareware. Try:
ftp.cica.indiana.edu, oak.oakland.edu or wuarchive.wustl.edu
Author: Dr. Shalom Halevy 70274.2564@compuserve.com PO BOX
22564, San Diego, CA 92192 (619) 552-9031 USA (Tel/FAX)
_________________________________________________________________
1.2.15 What is Shorten? Where can I get it?
Shorten is a compressor/coder for waveform files. Two major
changes have been made since the last announcement:
a) Thanks to the efforts of two users there is now a MS-DOS
executable (version 1.09) available on:
file://svr-ftp.eng.cam.ac.uk/comp.speech/sources/shn109.exe
b) The lastest version, 1.11, has early support for lossy
compresson. This is achieved by quantisation of the
prediction residual which maximises the segmental signal to
noise ratio. This works well for many waveforms - for
speech the quality is sometimes better and sometimes worse
than the various CCITT ADPCM standards. The advantages are
that the code is very fast, will accept most known file
formats and will code from lossless compression down to
three bits per sample. The disadvange is that this is a
variable bit rate scheme and so is more suited to storage
than transmission applications. It is available from:
file://svr-ftp.eng.cam.ac.uk/comp.speech/sources/shorten-1.11.tar.Z
The MS-DOS version comes with no support whatsoever - you
have been warned. I'll be able to test and maintain this
code when someone decides that it is worth funding the kit
to enable me to do this.
The UNIX version has been tested on many platforms and there
are no known portability problems. If you have problems,
then please tell me.
Feedback from USENET readers has been very valuable in the
past, and I'd like to ask for this again. I'll incorporate
as many sugestions as I can into version 2.0.
Contact: Tony Robinson (ajr@dsl.eng.cam.ac.uk)
Q2.1: Where can I get some algorithms for general DSP?
The following archives contain things such as matrix
operations, FFT's and generally useful things like that, as
opposed to complete applications.
_________________________________________________________________
Netlib, which serves some of this software via email. Try mail to
netlib@ORNL.GOV with "send help" in the subject field.
For Europe:
Internet: netlib@nac.no
EARN/BITNET: netlib%nac.no@norunix.bitnet
X.400: s=netlib; o=nac; c=no;
EUNET/uucp: nac!netlib
For the Pacific, try netlib@draci.cs.uow.edu.au
For background about netlib, see Jack J. Dongarra and Eric Grosse,
"Distribution of Mathematical Software Via Electronic Mail,"
Comm. ACM (1987) 30,403--407.
A similar collection of statistical software is available from
statlib@temper.stat.cmu.edu.
The symbolic algebra system REDUCE is supported by
reduce-netlib@rand.org.
_________________________________________________________________
The Naval Surface Warfare Center has a library of mathematical
Fortran subroutines that may be of use. From the report itself:
NSWC Library of Mathematical Subroutines
Report No.: NSWC TR 90-21, January 1990
by Alfred H. Morris, Jr.
Naval Surface Warfare Center (E43)
Dahlgren, VA 22448-5000
U.S.A.
Distribution: Approved for public release; distribution unlimited.
Abstract:
The NSWC library is a library of general-purpose Fortran
subroutines that provide a basic computational capability in
a variety of mathematical activities. Emphasis has been
placed on the transportability of the codes. Subroutines are
available in the following areas: Elementary Operations,
Geometry, Special Functions, Polynomials, Vectors, Matrices,
Large Dense Systems of Linear Equations, Banded Matrices,
Sparse Matrices, Eigenvalues and Eigenvectors, l1 Solution of
Linear Equations, Least-Squares Solution of Linear Equations,
Optimization, Transforms, Approximation of Functions, Curve
Fitting, Surface Fitting, Manifold Fitting, Numerical
Integration, Integral Equations, Ordinary Differential
Equations, Partial Differential Equations
[Witold Waldman, witold@hotblk.aed.dsto.gov.au]
This is avialble from
file://euler.math.usma.edu/pub/misc/nswc.tar.Z
This is a 3.2 Mbyte file with 800+ Fortran routines mentioned
above.
_________________________________________________________________
Also, you can get the Fortran source code from the IEEE Press book
"Programs For Digital Signal Processing." See question 1.2.
Also, see the summary of DSP-related FTP sites, at the end of
this FAQ.
If you don't know where to find what you're after, try archie.
_________________________________________________________________
SigLib
SigLib is an ANSI C Source DSP library. Current version is
1.61 SigLib has been compiled to run on IBM PCs, Sun
Workstations and the following DSPs : TMS320C30, TMS320C40,
DSP96002 and ADSP21020.
SigLib contains about 130 base functions, from which many
others are derived and over 80 demonstration programs, all of
which exercise more than on part of the library at any one
time. The library source and examples supplied total more
than 18000 lines of code. SigLib also includes DFilter, an
FIR and IIR digital filter design program and WinBuf, a
Windows 3 graphical front end, for display of process
results.
Some applications of SigLib include drill string vibration
analysis, room response analysis, audio effects,
telecommunications, active control of sound and vibration,
system simmulation and medical imaging.
Registered users of SigLib get one years free upgrade and
maintenance.
Spectrum analysis : FFTs and IFFTs; real, complex, zoom and
spectrograms, microscan. Windowing types : Hanning, Hamming,
Blackman, Triangle, Rectangle, Kaiser and Blackman-Harris.
Fixed coefficient filtering : FIR, comb, IIR and one pole IIR
filters, filter design methods, polyphase multi-rate filters,
differentiation and integration filters, Hilbert
transformers. Adaptive coefficient filtering : LMS.
Convolution and Correlation : convolve, correlate. Imaging :
conv3x3, histogram and 2DFFT. Signal generation : Sine,
Cosine, White noise, Chirp (linear and non-linear), Square,
Triangular, Sawtooth, Impulse, PN sequence. Modulation : AM,
complex shift, FSK, spectral inversion, FM, QAM. Statistical
analysis : sum, mean, average, standard deviation and
variance, kurtosis. Regression analysis : linear,
logarithmic, exponential, power. Digital effects : reverb,
distortion, echo, pitch shifting Utility functions, including
: scaling (lin and log) offset, min/max find, clip, offset,
rotate, buffer lengthen, buffer shorten, buffer addition
multiplication etc., histogram, quantise, absolute, peak
hold, polynomial expansion. Control : PID. Graphics :
display_buffer, display_buffer_line, print_buffer,
display_3d_buffer, pole_zero_plot, xy_plot. Data stream disk
I/O functions.
ITEMPRICE
SigLib object code UKP20, US$30
SigLib source and object code UKP35, US$60
Educational Price (Source code) UKP25, US$40
Available on 3.5" diskette
UK shipping per package UKP3
Non UK shipping per diskette UKP5, US$8
These fees include 1 years free upgrade and maintenance.
Payment preferably by Money Order or Cheque.
email:johned@cix.compulink.co.uk
John Edwards, Numerix, 157 Sileby Road, Barrow-on-Soar,
Leics, LE12 8LW, UK.
Phone : +44 (0)509 413195, UK time between 17.30 PM and 9.00
PM.
Q2.2: What are CELP and LPC? Where can I get the source for CELP and LPC?
CELP stands for "code excited linear prediction". LPC stands for
"linear predictive coding". They are compression algorithms used for
low bit rate (2400 and 4800 bps) speech coding. You can't get the
source for LPC anywhere on the net.
The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited linear
prediction voice coder version 3.2 (CELP 3.2) Fortran and C simulation
source codes are available for worldwide distribution (on DOS
diskettes, but configured to compile on Sun SPARC stations) from NTIS
and DTIC. Example input and processed speech files are included. A
Technical Information Bulletin (TIB), "Details to Assist in
Implementation of Federal Standard 1016 CELP," and the official
standard, "Federal Standard 1016, Telecommunications: Analog to
Digital Conversion of Radio Voice by 4,800 bit/second Code Excited
Linear Prediction (CELP)," are also available.
This is available through the National Technical Information Service:
NTIS
U.S. Department of Commerce
5285 Port Royal Road
Springfield, VA 22161
USA
(703) 487-4650
It may also be obtained from:
file://svr-ftp.eng.cam.ac.uk/comp.speech/sources/DoD_CELP-3.2.tar.Z
An updated version of the FS-1016 CELP 3.2 code is available, at least
for a while, on:
file://super.org/pub/celp_3.2a.tar.Z
The code (C, FORTRAN, diskio) all has been built and tested on a Sun4
under SunOS4.1.3. If you want to run it somewhere else, then you may
have to do a bit of work. (A Solaris 2.x-compatible release is
planned soon.)
[One note to PCers. The files:
[
[ cbsearch.F celp.F csub.F mexcite.F psearch.F
[
[are meant to be passed through the C preprocessor (cpp).
[We gather that DOS (or whatever it's called) can't distinguish
[the .F from a .f. Be careful!
Very limited support is available from the authors (Joe, et al.).
Please do not send questions or suggestions without first reading the
documentation (README files, the Technical Information Bulletin, etc.).
The authors would enjoy hearing from you, but they have limited time
for support and would like to use it as efficiently as possible. They
welcome bug reports, but, again, please read the documentation first.
All users of FS-1016 CELP software are strongly encouraged to acquire
the latest release (version 3.2a as of this writing).
The "AD" ordering number for the CELP software is AD M000 118
(US$ 90.00) and for the TIB it's AD A256 629 (US$ 17.50). The LPC-10
standard, described below, is FIPS Pub 137 (US$ 12.50). There is a
$3.00 shipping charge on all U.S. orders. The telephone number for
their automated system is 703-487-4650, or 703-487-4600 if you'd prefer
to talk with a real person.
(U.S. DoD personnel and contractors can receive the package from the
Defense Technical Information Center: DTIC, Building 5, Cameron
Station, Alexandria, VA 22304-6145. Their telephone number is
703-274-7633.)
The following articles describe the Federal-Standard-1016 4.8-kbps CELP
coder (it's unnecessary to read more than one):
Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch,
"The Federal Standard 1016 4800 bps CELP Voice Coder," Digital Signal
Processing, Academic Press, 1991, Vol. 1, No. 3, p. 145-155.
Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch,
"The DoD 4.8 kbps Standard (Proposed Federal Standard 1016),"
in Advances in Speech Coding, ed. Atal, Cuperman and Gersho,
Kluwer Academic Publishers, 1991, Chapter 12, p. 121-133.
Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
Proposed Federal Standard 1016 4800 bps Voice Coder: CELP," Speech
Technology Magazine, April/May 1990, p. 58-64.
The U.S. DoD's Federal-Standard-1015/NATO-STANAG-4198 based 2400 bps
linear prediction coder version 53 (LPC-10e v53) Fortran or C simulation
source codes are available on a limited basis upon written request to:
Tom Tremain
Department of Defense
Ft. Meade, MD 20755-6000
USA
There is also a section about FS-1015 in the book:
Panos E. Papamichalis, Practical Approaches to Speech Coding,
Prentice-Hall, 1987.
The following article describes the FS 1016 4.8-kbps CELP coder:
Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
Proposed Federal Standard 1016 4800 bps Voice Coder: CELP," Speech
Technology Magazine, April/May 1990, p. 58-64.
Copies of the official standard
"Federal Standard 1016, Telecommunications: Analog to Digital Conversion
of Radio Voice by 4,800 bit/second Code Excited Linear Prediction (CELP)"
are available for US$ 5.00 each from:
GSA Federal Supply Service Bureau
Specification Section, Suite 8100
470 E. L'Enfant Place, S.W.
Washington, DC 20407
(202)755-0325
The U.S. Federal Standard 1015 (NATO STANAG 4198) is described in:
Thomas E. Tremain, "The Government Standard Linear Predictive Coding
Algorithm: LPC-10," Speech Technology Magazine, April 1982, p. 40-49.
The voicing classifier used in the enhanced LPC-10 (LPC-10e) is described in:
Campbell, Joseph P., Jr. and T. E. Tremain, "Voiced/Unvoiced Classification
of Speech with Applications to the U.S. Government LPC-10E Algorithm,"
Proceedings of the IEEE International Conference on Acoustics, Speech, and
Signal Processing, 1986, p. 473-6.
Realtime DSP code for FS-1015 and FS-1016 is sold by several vendors,
including DSP Software Engineering and Analogical Systems (see the
vendor address list in section 5 for contact info). DSP Software
Engineering's FS-1016 code can run on a DSP Research's Tiger 30 or on
Intellibit's AE2000 TMS320C31 based 3" by 2.5" card. See section 4.1
for more on these cards. Analogical's product runs on a 27 MHz
DSP56001 chip.
[Most of the above from Joe Campbell, jpcampb@afterlife.ncsc.mil, with
additions from Dan Frankowski, drankow@cs.umn.edu, and Ed Hall,
edhall@rand.org]
Q2.3: What is ADPCM? Where can I get source for it?
ADPCM stands for Adaptive Differential Pulse Code Modulation. It is a
family of speech compression and decompression algorithms. A common
implementation takes 16-bit linear PCM samples samples and converts
them to 4-bit samples, yeilding a compression rate of 4:1.
There is public domain C code available via anonymous ftp at
file://ftp.cwi.nl/pub/audio/adpcm.shar written by Jack Jansen (email
Jack.Jansen@cwi.nl). It is very programmer-friendly. The ADPCM code
used is the Intel/DVI ADPCM code which is being recommended by the IMA
Digital Audio Technical Working Group. It allows the following calls:
adpcm_coder(short inbuf[], char outbuf[], int nsample,
struct adpcm_state *state);
adpcm_decoder(char inbuf[], short outbuf[], int nsample,
struct adpcm_state *state);
The routines have been tested on an SGI Indigo running Irix 4.0.2 and
on a Sparcstation 1+ running SunOS 4.1.1. On a Sun, the code will
compress at 250Ksample/sec and decompress at 300Ksample/sec. On an
SGI, the compressor runs at 350Ksample/sec and the decompressor at
700Ksample/sec.
Note that this is NOT a CCITT G722 coder. The CCITT ADPCM standard is
much more complicated, probably resulting in better quality sound but
also in much more computational overhead.
You can get a G.721/722/723 package by email to teledoc@itu.arcom.ch, with
GET ITU-3022
as the *only* line in the body of the message.
This is also available as:
file://svr-ftp.eng.cam.ac.uk/comp.speech/sources/G711_G722_G723.tar.Z
[From Dan Frankowski, drankow@cs.umn.edu; Jack Jansen, Jack.Jansen@cwi.nl]
Q2.4: What is GSM? Where can I get source for it?
The README file for GSM says:
GSM 06.10 13 kbit/s RPE/LTP speech compression available
--------------------------------------------------------
The Communications and Operating Systems Research Group (KBS) at the
Technische Universitaet Berlin is currently working on a set of
UNIX-based tools for computer-mediated telecooperation that will be
made freely available.
As part of this effort we are publishing an implementation of the
European GSM 06.10 provisional standard for full-rate speech
transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse
excitation/long term prediction) coding at 13 kbit/s.
GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
with typical UNIX applications, our implementation turns frames of 160
16-bit linear samples into 33-byte frames (1650 Bytes/s).
The quality of the algorithm is good enough for reliable speaker
recognition; even music often survives transcoding in recognizable
form (given the bandwidth limitations of 8 kHz sampling rate).
The interfaces offered are a front end modelled after compress(1), and
a library API. Compression and decompression run faster than realtime
on most SPARCstations. The implementation has been verified against the
ETSI standard test patterns.
Jutta Degener (jutta@cs.tu-berlin.de)
Carsten Bormann (cabo@cs.tu-berlin.de)
Communications and Operating Systems Research Group, TU Berlin
Fax: +49.30.31425156, Phone: +49.30.31424315
An implementation can be had from:
file://tub.cs.tu-berlin.de/pub/tubmik/gsm-1.0.tar.Z
with file://tub.cs.tu-berlin.de/pub/tubmik/gsm-1.0-patch1
and file://tub.cs.tu-berlin.de/pub/tubmik/gsm-1.0-patch2
or as a faster but not always up-to-date alternative:
file://liasun3.epfl.ch/pub/audio/gsm-1.0pl1.tar.Z
[From Dan Frankowski, dfrankow@cs.umn.edu]
Q2.5: How does pitch perception work, and how do I implement it on my DSP chip?
Pitch is officially defined as "That attribute of auditory sensation
in terms of which sounds may be ordered on a musical scale." Several
good examples illustrating the subtleties of pitch perception are
included in the "Auditory Demonstrations CD" which is available from
the Acoustical Society of America, Woodbury, NY 10797 for $20.
A good general reference about the psychology of pitch perception is
the book:
B.C.J. Moore, "An Introduction to the Psychology of Hearing",
Academic Press, London, 1989.
This book is available in paperback and makes a good desk reference.
An algorithm implementation that matches a large body of psychoacoustical
work, but which is computationally very intensive, is presented in the paper:
Malcolm Slaney and Richard Lyon, "A Perceptual Pitch Detector,"
Proceedings of the International Conference of Acoustics, Speech,
and Signal Processing, 1990, Albuquerque, New Mexico.
The definitive papers describing the use of such a perceptual pitch
detector as applied to the classical pitch literature is in:
Ray Meddis and M. J. Hewitt. "Virtual pitch and phase
sensitivity of a computer model of the auditory periphery. "
Journal of the Acoustical Society of America 89 (6 1991): 2866-2682.
and 2883-2894.
The current work that argues for a pure spectral method starts with the work
of Goldstein:
J. Goldstein, "An optimum processor theory for the
central formation of the pitch of complex tones," Journal
of the Acoustical Society of America 54, 1496-1516, 1973.
Two approaches are worth considering if something approximating pitch
is appropriate. The people at IRCAM have proposed a harmonic analysis
approach that can be implemented on a DSP
Boris Doval and Xavier Rodet, "Estimation of Fundamental Frequency
of Musical Sound Signals," Proceedings of the 1991 International
Conference on Acoustics, Speech, and Signal Processing, Toronto,
Volume 5, pp. 3657-3660.
The classic paper for time domain (peak picking) pitch algorithms is:
B. Gold and L. Rabiner, "Parallel processing techniques for estimating
pitch periods of speech in the time domain," Journal of the Acoustical
Society of America, 46, pp 441-448, 1969.
Finally, a word of caution: Pitch is not single-valued. We can hear a
sound and match it to several different pitches. Imagine the number
of instruments in an orchestra, each with its own pitch. Even a
single sound can have more than one pitch. See for example
Demonstration 27 from the ASA Auditory Demonstrations CD.
[The above from Malcolm Slaney, Apple Computer, and John Lazzaro,
U.C. Berkeley.]
Q2.6: What standards are there for digital audio? What is AES/EBU?
What is S/P-DIF?
The "AES/EBU" (Audio Engineering Society / European Broadcast Union)
digital audio standard is probably the most popular digital audio
standard today. Most consumer and professional digital audio devices
(CD players, DAT decks, etc.) that feature digital audio I/O support
AES/EBU.
AES/EBU is a bit-serial communications protocol for transmitting
digital audio data through a single transmission line. It provides two
channels of audio data (up to 24 bits per sample), a method for
communication control and status information ("channel status bits"),
and some error detection capabilities. Clocking information (i.e.,
sample rate) is derived from the AES/EBU bit stream, and is thus
controlled by the transmitter. The standard mandates use of 32 kHz,
44.1 kHz, or 48 kHz sample rates, but some interfaces can be made to
work at other sample rates.
AES/EBU provides both "professional" and "consumer" modes. The big
difference is in the format of the channel status bits mentioned above.
The professional mode bits include alphanumeric channel origin and
destination data, time of day codes, sample number codes, word length,
and other goodies. The consumer mode bits have much less information,
but do include information on copy protection (naturally). Additionally,
the standard provides for "user data", which is a bit stream containing
user-defined (i.e., manufacturer-defined) data. According to Tim
Channon, "CD user data is almost raq CD subcode; DAT is StartID and
SkipID. In progfessional mode, there is an SDLC protocol or, if DAT,
it may be the same as consumer mode."
The physical connection media are commonly used with AES/EBU:
balanced (differential), using two wires and shield in three-wire microphone
cable with XLR connectors; unbalanced (single-ended), using audio coax cable
with RCA jacks; and optical (via fiber optics).
"S/P-DIF" (Sony/Philips Digital Interface Format) typically refers to
AES/EBU operated in consumer mode over unbalanced RCA cable. Note
that S/P-DIF and AES/EBU mean different things depending on how much
of a purist you are in the digital audio world; see the Finger article
below.
References:
Finger, Robert, "AES3-199X: The Revised Two Channel Digital Audio
Interface (DRAFT)", presented at the 91st Convention of the Audio
Engineering Society, October 4-8, 1991. Reprints: AES, 60 East 42nd
St., New York, NY, 10165.
[The above from Phil Lapsley, phil@ohm.Berkeley.EDU, and Tim Channon,
tchannon@black.demon.co.uk]
Q2.7: What is mu-law encoding? Where can I get source for it?
Mu-law (also "u-law") encoding is a form of logarithmic quantization
or companding. It's based on the observation that many signals are
statistically more likely to be near a low signal level than a high
signal level. Therefore, it makes more sense to have more quantization
points near a low level than a high level. In a typical mu-law system,
linear samples of 14 to 16 bits are companded to 8 bits. Most telephone
quality codecs (including the Sparcstation's audio codec) use mu-law
encoded samples.
Desktop Sparc machines come with routines to convert between linear and
mu-law samples. On a desktop Sparc, see the man page for audio_ulaw2linear
in /usr/demo/SOUND/man.
Craig Reese posted the source of similar routines to comp.dsp in August '92.
These are archived on file://evans.ee.adfa.oz.au/pub/dsp/misc
References:
CCITT Recommendation G.711 (very difficult to follow).
Michael Villeret, et. al, "A New Digital Technique for Implementation
of Any Continuous PCM Companding Law,", IEEE Int. Conf. on Communications,
1973, vol. 1, pp. 11.12-11.17.
MIL-STD-188-113, "Interoperability and Performance Standards
for Analog-to-Digital Conversion Techniques," 17 February 1987.
"TI Digital Signal Processing Applications with the TMS320 Family",
pp. 169-198.
[From Joe Campbell; Craig Reese, cfreese@super.org; Sepehr Mehrabanzad,
sepehr@falstaff.dev.cdx.mot.com]
Q2.8: How can I do CD DAT sample rate conversion?
CD players use a 44.1 kHz sample rate, whereas DAT uses a 48
kHz sample rate. This means that you must do sample rate
conversion before you can get data from a CD player directly
into a DAT deck.
[From Ed Hall, edhall@rand.org:]
For a start, look at "Multirate Digital Signal Processing"
by Crochiere and Rabiner (see FAQ section 1.1).
Almost any technique for producing good digital low-pass
filters will be adaptable to sample-rate conversion. 44.1:48
and vice-versa is pretty hairy, though, because the lowest
whole-number ratio is 147:160. To do all that in one go
would require a FIR with thousands of coefficients, of which
only 1/147th or 1/160th are used for each sample--the real
problem is memory, not CPU for most DSP chips. You could
chain several interpolators and decimators, as suggested by
factoring the ratio into 3*7*7:2*2*2*2*2*5. This adds
complexity, but reduces the number of coefficients required
by a considerable amount.
__________________________________________________________________________
[From Lou Scheffer:]
Theory of operation: 44.1 and 48 are in the ratio 147/160.
To convert from 44.1 to 48, for example, we (conceptually):
1) interpolate 159 zeros between every input sample. This
raises that data rate to 7.056 MHz. Since it is
equivalent to reconstructing with delta functions, it
also creates images of frequency f at 44.1-f, 44.1+f,
88.2-f, 88.2+f, ...
2) We remove these with an FIR digital filter, leaving a
signal containing only 0-20 KHz information, but still
sampled at a rate of 7.056 MHz.
3) We discard 146 of every 147 output samples. It does
not hurt to do so since we have no content above 24 KHz.
In practice, of course, we never compute the values of
the samples we will throw out.
So we need to design an FIR filter that is flat to 20 KHz,
and down at least X db at 24 KHz. How big does X need to
be? You might think about 100 db, since the max signal size
is roughly +-32767, and the input quantization +- 1/2, so we
know the input had a signal to broadband noise ratio of 98
db at most. However, the noise in the stopband
(20KHz-3.5MHz) is all folded into the passband by the
decimation in step 3, so we need another 22 db (that's 160
in db) to account for the noise folding. Thus 120 db
rejection yields a broadband noise equal to the original
quantizing noise. If you are a fanatic, you can shoot for
130 db to make the original quantizing errors dominate, and
a 22.05 KHz cutoff to eliminate even ultrasonic aliasing.
You will pay for your fanaticism with a penance of more
taps, however.
__________________________________________________________________________
For more details, a technical report (the author's name is
missing, if you know who - let me know!), is available is
LaTeX source form as
file://evans.ee.adfa.oz.au/pub/dsp/cd-rate-convert.tex
and in PostScript as
file://evans.ee.adfa.oz.au/pub/dsp/cd-rate-convert.ps
__________________________________________________________________________
There's a free implementation of Julius O. Smith III and someone
else's "bandwidth-limited interpolation" rate conversion algorithm.
The paper available as
file://ccrma-ftp.stanford.edu/pub/DSP/Tutorials/BandlimitedInterpolation.eps.Z
explains the algorithm. The source code in
file://netcom.com/pub/thinman/resample.01.Z
file://netcom.com/pub/thinman/resample.02.Z
file://netcom.com/pub/thinman/kaiser.c
implements the algorithm. It all works quite well.
--
-Kabir-
----------------------------------------------------------------------
Path: news1.ucsd.edu!ihnp4.ucsd.edu!swrinde!howland.reston.ans.net!newsfeed.internetmci.com!in1.uu.net!usc!sdd.hp.com!hplabs!unix.sri.com!news.Stanford.EDU!newshub.internex.net!news.internex.net!tang.bdti.com!kabir
From: phil@bdti.com
Newsgroups: comp.dsp,comp.answers,news.answers
Subject: comp.dsp FAQ [2 of 3]
Supersedes: <compdsp.2_818463006@bdti.com>
Followup-To: poster
Date: 20 Dec 1995 00:59:41 GMT
Organization: Berkeley Design Technology, Inc.
Lines: 1692
Approved: news-answers-request@MIT.Edu
Distribution: world
Expires: 2 Feb 1996 01:04:40 GMT
Message-ID: <compdsp.2_819421480@bdti.com>
References: <compdsp.1_819421480@bdti.com>
NNTP-Posting-Host: bdti.com
Summary: This is a periodic posting to comp.dsp that gives information
on frequently asked questions asked in this newsgroup.
Originator: kabir@tang.bdti.com
Xref: news1.ucsd.edu comp.dsp:20124 comp.answers:13193 news.answers:51482
Archive-name: dsp-faq/part2
Last-modified: Wed June 21 1995
Version: 2.2
Q3.1: What are the available DSP chips and chip architectures?
{ This is based on a woefully inadequate databook collection. Anyone want
to add to this list? Manufacturers want to submit anything? }
The "big four" programmable DSP chip manufacturers are Texas Instruments,
with the TMS320 series of chips; Motorola, with the DSP56000 and DSP96000
series; AT&T, with the DSP16 and DSP32 series; and Analog Devices, with
the ADSP2100 series. A good overview of prorammable DSP chips is published
periodically in EDN magazine. The most recent version is from Sep. 17, 1992,
pp. 90-141.
Here's a less ambitious chip breakdown by manufacturer:
_________________________________________________________________
Texas Instuments:
TMS320C1x: family of low cost fixed-point DSP's; 16 bit data, 32 bit
registers; Various RAM and ROM configurations; 16 bit I/O bus, serial ports.
TMS320C25: 50MHz fixed-point DSP; 16 bit data, 32 bit registers;
12.5 MIPS @ 50MHz.
TMS320C30: 27/33/40 MHz floating point DSP; 32 bit floating point,
24 bit fixed point data, 40 bit registers; DMA controller; dual serial
ports; some support for multi-processor arrays.
TMS320C31: version of C30 minus peripheral bus, one serial port, and the 4Kx32
internal ROM. ~$20, 132 pin PQFP.
TMS320C40: 40/50 MHz floating point DSP; extensive parallel processing
support through 6 buffered byte-wide 20 Mb/s links and 6 channel DMA; cache.
TMS320C50: enhanced TMS320C25 (double throughput); low overhead looping;
10 Kwords SRAM on chip.
TMS320C80: four XXX MHz fixed-point processors combined with a RISC
supervisory processor in a single multichip module.
_________________________________________________________________
Motorola:
DSP56001: 20.5, 27, or 33 MHz 24-bit fixed point DSP. 24 bit data bus, 16 bit
address bus, 56 bit accumulators (2), host interface port, serial ports (2),
general purpose I/O pins, timer. Harvard architecture. 512 words program
RAM, 32 words bootstrap ROM, 512 words data RAM, 512 words data ROM on chip.
Available in PGA, CQFP or PQFP packaging.
DSP56000: Mask-programmed version of DSP56001, same peripherals and data
memories, 3.75k words program ROM on chip.
DSP56002: modular DSP based on new 24-bit DSP56k core, a superset of
the DSP56001 architecture with On-Chip Emulation (OnCE) debug port,
clock PLL and improved bus arbitration. Has four cycle double precision
multiply and support for block floating point. Same memory as in
DSP56001, except for 64 words bootstrap ROM. Host interface port,
serial ports (2), general purpose I/O pins, programmable 24-bit timer,
non-maskable interrupt. Low power fully static design, no minimum
clock frequency requirement. Available at 40 MHz (5V supply) in PGA
and CQFP packaging.
DSP56004: modular DSP, same 24-bit DSP56k core as in DSP56002. Targeted
to consumer digital audio applications. Has On-Chip Emulation (OnCE)
debug port, clock PLL, serial host interface (I2C and SPI), four
general purpose I/O pins, two stereo serial audio receivers (I2S/Sony),
three stereo serial audio transmitters (I2S/Sony), external SRAM/DRAM
memory interface with 8-bit data bus. Low power fully static design,
no minimum clock frequency requirement. Available at 40 MHz (5V
supply) in 80-pin QFP package.
DSP56156: 40, 50, or 60 MHz fixed point DSP; 16 bit data bus, 40 bit
accumulators (2), host interface port, serial ports (2), timer, OnCE
debug port, clock PLL, 14 bit sigma-delta voice band CODEC, 2K words
program RAM, 2K words data RAM on chip.
DSP96002: IEEE format floating point DSP; two complete 32 bit data and
address buses; Harvard architecture. 1k words program RAM, 64 words
bootstrap ROM, 1k words data RAM, 1k words data ROM, host interface
ports (2). Available in 33 MHz or 40 MHz in 223-pin PGA packaging.
[The above from Sergio Liberman, sergio@msil.sps.mot.com ]
_________________________________________________________________
AT&T:
DSP 16 FAMILY: DSP16A, $22.60; DSP1610, $91; DSP1616, $35.70
16 bit fixed point DSPs. The DSP16A has a 25ns cycle time while the
16C has an ADC and DAC on chip. The C-version also has a 4-pin JTAG
interface. The 1610/1616 are enhanced versions intended for cellular
phone use. The chips use separate on-chip 16-bit program- and data
buses. The A and C versions have 12k x 16-bit program ROM and a 2k x
16-bit data-RAM while featuring parallell and serial I/O.
DSP 32C/3210: DSP32C, $70(1000); DSP3210, $50(100k)
32-bit floating point with 40-bits accumulator and 16/24-bit fixed
point. These DSP's uses a single 4M-word (3210: 4G-word) linear memory
space instead of the separate program and data memory found on most
DSP:s. The bus can be accessed four times and each internal memory two
times per cycle. The 3210, along with the VCOS operating system is
intended for use on the mother board of personal computers and
workstations where it shares memory with the host. The 32C has three
512 x 32-bit RAM:s while the 3210 has two 1k x 32-bit RAMs and a 256 x
32-bit boot ROM. 32C: Serial and parallell I/O, 3210: Serial I/O,
timer, DMA-controller. 3210 available at 50 and 66MHz.
DSP3210/DSP3207:
32-bit floating point with four 40-bit floating-point accumulators
and twenty-two general purpose 32-bit fixed point registers.
Single 32-bit (4G-byte) linear memory space. Support for
byte, 16-bit word, and 32-bit word accesses. Big/little
endian interface. C-like assembly language. Up to four memory
accesses per instruction. VCOS operating system allows sharing
of host memory (e.g. mother-board or local-bus board) or operation
out of inexpensive DRAM (e.g. ISA board). Two 1k x 32-bit RAMs
and a 256 x 32-bit boot ROM. Serial I/O, timer, DMA-controller.
The DSP3207 is functionally equivalent to the DSP3210 except it lacks
the serial I/O and associated DMA controller and has been designed
for low power dissipation.
DSP3210 available at 55MHz/5V and 66MHz/5V.
DSP3207 available at 55MHz/5V and 66MHz/5V and 40MHz/3V.
_________________________________________________________________
Analog Devices:
ADSP-2100: 40 and 50Mhz fixed point DSP (10 MIPS, 12.5MIPS). 16 bit
operations with 40-bit multiply-accumulate. No on chip memory except for a 16
word instruction cache. Off chip harvard architecture. PGA and PQFP
packages.
ADSP-2101: Derived from ADSP-2100; 16 bit operations with 40-bit MAC
register. Extras include on chip memory of 2Kx24 program memory
instruction/data RAM and 1Kx16 data memory RAM, 16-bit timer, two serial
ports, low power state. PGA, PLCC, PQFP packages. Fastest speed grade in
production is 16.6 MHz (16.6MIPS).
ADSP-2102: RAM/ROM version of 2101; user selects how much of the 2kx24
program memory is mask ROM.
ADSP-2103: 3.3V version of the 2101 running at 13MHz (13 MIPs). PLCC, PQFP
packages.
ADSP-2105: 10Mhz (10MIPS) low cost fixed point DSP with 1 serial port, timer
and 1kx24 instruction/data RAM in program memory space, and 512 word data RAM
in data memory space. Architecture and instruction set identical to
ADSP-2101. Pin compatible with 2101. PLCC package only (can use standard 68
pin plcc socket). This processor sells for US $9.90 in any quantity.
ADSP-2115: Architecture and pinout same as ADSP-2101, but 1K program memory
RAM, 1/2K data memory RAM with 2 serial ports, interval timer etc. PLCC and
PQFP packages. Available in 13.8MHz and 16MHz (13.8, 16 MIPs)
ADSP-2111: Adds a 8/16bit host interface port to ADSP-2101 architecure
allowing interface to Intel or Motorola style microprocessors. 13 and 16 MIPs
speeds available. PGA and PQFP packages.
ADSP-21msp50: ADSP-2111 with an on chip a/d and d/a interface (65dB SNR)
Additional low power modes allow CMOS standby (sors include hardware support fo
r zero overhead looping, modulo
addressing, single cycle context switch, and bit reversal addressing. All
instructions, even those which access external memory can complete in 1 cycle.
[Analog Devices DSP applications, dsp_applications@analog.com]
Q3.2: Software for Motorola DSPs.
_________________________________________________________________
Q3.2.1: Where can I get a free assembler for the Motorola DSP56000?
A free assembler for the Motorola DSP56000 exists, thanks to Quinn
Jensen, jensenq@qcj.icon.com. The current version is 1.1, and it is
posted to alt.sources, so look for it on mirrors of that newsgroup
(like wuarchive.wustl.edu).
_________________________________________________________________
Q3.2.2: Where can I get a free C compiler for the Motorola DSP56000?
There are two separate compiler sources for the Motorola DSP56000. One
is the port of gcc 1.40 done by Andrew Sterian
(asterian@eecs.umich.edu) and the other is a port of gcc 1.37.1 done
by Motorola and returned to the FSF. Andrew's port has bowed to
Motorola's version. Both may be portable to gcc2.x.x with some effort
required. Neither of these comes with an assembler, but you can get a
free DSP56000 assembler elsewhere (see Q3.2.1 above). The Motorola gcc
source is available for FTP from:
Site Directory
nic.funet.fi ~pub/ham/dsp/dsp56k-tools/dsp56k-gcc.tar.Z
evans.ee.adfa.oz.au pub/micros/56k/g56k.tar.Z
From Andrew Sterian, asterian@eecs.umich.edu: "My DSP56156 port is
still the only DSP56156 compiler around and I have just released an
updated version of it. Both this compiler and the previous incarnation
are archived on wuarchive.wustl.edu (in the usenet/alt.sources
directory) amongst other places."
_________________________________________________________________
Q3.2.3: Where can I get algorithms and libraries for Motorola DSPs?
What is the number for the Motorola DSP BBS?
Motorola runs "Dr. Bub", a bulletin board for DSPs containing source
code for various libraries and algorithms. You can call it at (512)
891-3771 (9600, 4800, 2400, 1200 bps) or (512) 891-3773 (2400/1200/300
bps). Format is 8 data bits, no parity, 1 stop bit). Log in as "guest"
to browse the system, or you can open an account by entering "new" at
the account name prompt. [John Fisher, johnf@dsp.sps.mot.com]
Alternatively, Dr. BuB is mirrored on the following sites:
calvin.stanford.edu (36.14.0.43) in /motorola bode.ee.
ualberta.ca (129.128.16.96) in /pub/dos/motorola
nic.funet.fi (128.214.6.100) in /pub/misc/motorola
doc.ic.ac.uk (146.169.3.7)
in /computing/systems/motorola/digital-signal-processing/dr.bub.sources
Also try nic.funet.fi in /pub/ham/dsp for a lot of good stuff on
communications uses, including some hardware.
ccrma-ftp.stanford.edu also has a variety of DSP code (much of it NeXT
specific, see below), including the following for the DSP56000:
pub/clm.tar.Z "CLM", a package aimed mainly at composers doing
computer music in Common Lisp, but includes a Lisp 56000 assembler,
debugger, loader, large libraries of DSP56000 routines useful in
computer music, and a compiler from a subset of Common Lisp to
DSP56000 code. [bil@ccrma.stanford.edu]
_________________________________________________________________
Q3.2.4: Where can I get NeXT-compatible Motorola DSP56001 code?
Try the following from ccrma-ftp.Stanford.EDU:
* DSP programs for the NeXT platform:
pub/DSP/resample.tar.Z
Audio sampling-rate conversion and FIR filter design.
pub/DSP/ResoLab2.1.tar.Z
Interactive filter instrument; sources now included, online
help.
pub/DSP/Spectro.Z
Spectrum analysis tool, with source code.
pub/DSP/WaveFormEditor.tar.Z
Jean Laroche's real-time waveform editor, with DAJ's additions.
* DSP programming examples for the NeXT platform:
pub/DSP/dsp_dma_stream.tar.Z
Fast DSP DMA programming example (two-way DMA).
pub/DSP/JeanLaroche.tar.Z
Low-level sound and DSP programming examples and docs.
[bil@ccrma.Stanford.EDU]
Q3.3: Software for Texas Instruments DSPs.
_________________________________________________________________
Q3.3.1: Where can I get free algorithms or libraries for TI DSPs?
What is the number for the TI DSP BBS?
nic.funet.fi has some old, apparently public domain, assembler and
related tools from TI for the TMS320 family. [Antti-Pekka Virtanen,
antsu@utu.fu]
The TI DSP bulletin board is at (713) 274-2323 (300, 1200, 2400, or
9600 bps; 8 data, 1 stop, no parity).
The TI DSP bulletin board is mirrored on ti.com, and on
evans.ee.adfa.oz.au. The TI site is the official one, but has no user
contributed software. The file:
file://evans.ee.adfa.oz.au/mirrors/tibbs/00README provides further
guidance. Please restrict FTP session to outside of 8 am to 6 pm local
time (10 pm to 8 am GMT). [Brad Hards, bradh@ee.adfa.oz.au]
{ If anyone knows of any other sources for TI DSP software, please let
us know at comp-dsp-faq@ohm.Berkeley.EDU. Thanks! }
_________________________________________________________________
Q3.3.2: Where can I get a free C compiler for the TI TMS320C30?
Sonitech (see vendors list) has a gcc based TMS320C30 C compiler that
was originally done Computer Motion. Sonitech sells it for $995, but
under the terms of the Gnu Public License, other people can then give
it away. While we haven't heard of any ftp sites yet, there are bound
to be some soon.
_________________________________________________________________
Q3.3.3: Where can I get a free assembler for the TI TMS320C30?
Ted Rossin has written an assembler and linker for the TMS320C30. In
his words, "It is somewhat limited by the fact that it can't handle
expressions but it has worked fine for me over the past few years.
There is no manual because it is a clone of the TI assembler and
linker. However the linker command files use a different (easier to
use) syntax. It runs on HP-UX workstations, Macs, IBM clones and
believe it or not the Atari-ST (because I developed the code on it)."
It is available for anonymous ftp from:
file://schutz.ee.uts.edu/pub/DSP/c30/as30.tar.Z [Ted Rossin,
rossin@fc.hp.com]
_________________________________________________________________
Q3.3.4: What is Tick? Where can I get it?
Tick is a TMS320C40 parallel network detection and loader utility.
It is available from:
file://evans.ee.adfa.oz.au/mirrors/tibbs/UserContributed
Supports: Transtech, Hunt, and Traquair boards hosted by DOS, SunOS,
Linux (a PC unix)
Q3.4: Software for Analog Devices DSPs.
_________________________________________________________________
Q3.4.1: Where can I get algorithms or libraries for Analog Devices
DSPs? What is the number for the Analog Devices DSP BBS?
The number for the Analog Devices DSP BBS is (617) 461-4258 (300,
1200,
2400, 9600, 14400 bps), 8N1.
[Analog Devices DSP Applications, dsp_applications@analog.com]
{ If anyone knows of other sources for Analog Devices DSP software,
please let us know at comp-dsp-faq@bdti.com. Thanks! }
Q4.1: DSP development boards.
Note: This information was mainly supplied by vendor catalogues. It is
in no way definitive, and much of the information may well be out of
date or simply wrong. Beware!
_________________________________________________________________
Q4.1.1: IBM PC DSP development boards.
IBM PC boards, Analog Devices 2100 series processors:
----------------------------------------------------------------------------
Name: ADSP-2101 DSPB 2101-4
Type: Four processor ADSP-2101 IBM PC AT board
Company: CMS GmbH
Processor: 4x ADSP 2101
Analog I/O: 2 16 bit A/Ds per processor.
Name: ADSP-2101 LAB-DSP
Type: ADSP-2101 or ADSP-2105 IBM PC/AT XT
Company: Computer Continuum
Processor: ADSP 2101 or ADSP 2105
Features: Shared memory between PC and DSP. Serial I/O. Daughter board.
Name: Feature Finder ADSP-2105
Type: ADSP-2105 frame grabber for IBM PC
Company: Current Technology, Inc.
Processor: ADSP 2101
Features: RS-170 video interface, 512x512 video memory, feature extraction s/w
Name: LAB2105 DSP Card
Type: ADSP-2105 board for IBM PC
Company: EnterTec, Inc.
Processor: ADSP-2105
Analog I/O: Codec with microphone input and headphone output.
Name: ADSP-2105 DSP Platform
Type: Dual ADSP-2105 or ADSP-2101 IBM PC board
Company: Hollis Electronics
Processor: 2x ADSP-2105 (or ADSP-2101s)
Analog I/O: 2 14 bit D/A, 2 12 bit A/D
Features: 2x RS-232 UARTs, real-time clock, 2x 16 bit timers, 4 char LED
Name: ADSP-2100 DX2100
Type: ADSP 2100 (?) IBM PC board
Company: Logabex
Processor: ADSP-2100 (?)
Name: ADSP-2100 System board
Type: ADSP-2100A IBM PC card
Company: Loughborough Sound Images
Processor: ADSP-2100A 40 MHz 16K words RAM
Analog I/O: 12 bit 125 kHz A/D, D/A
Name: ADSP-2101 System board
Type: ADSP-2101 IBM PC card
Company: Loughborough Sound Images
Processor: ADSP-2101 12.5 MHz, 8K words RAM
Analog I/O: A/D -- D/A: dual 14 bit 19.2 kHz CODECs (TI TLC32044Cs)
Comments: 32Kx8 EPROM socket allows some standalone behavior
Name: GODSPEED (ADSP-2101)
Type: ADSP-2101 IBM PC board
Company: Prime Ideal
Processor: ADSP-2101, dual-port RAM to PC bus
Analog I/O: 48 kHz codec w/ speaker and mic connectors
Name: ADSP-2105 Espresso Board
Type: ADSP-2105 IBM PC XT/AT board
Company: Saddle Point Systems
Processor: ADSP-2105 or ADSP-2101, 28K words memory.
Analog I/O: on board codec
Name: ADSP-2101 SPB2
Type: Dual ADSP-2101 IBM PC AT board
Company: Signal-Data
Processor: 2x ADSP-2101s, 32Kw RAM/proc, 8Kw ROM/proc.
Analog I/O: 12 bit A/D, 12 bit D/A on one processor
Name: PC-1601A, DSPS-2601
Type: ADSP-2101 IBM PC board (?)
Company: Wavetron Microsystems
Comments: DSPS-2601 is a multi-channel version. (?)
_________________________________________________________________
IBM PC boards, Analog Devices 21000 series processors:
----------------------------------------------------------------------------
Name: GAMMA 20/25
Type: ADSP-21020 IBM PC AT board
Company: BittWare Research Systems, Inc.
Processor: DSP-21020 25 MHz, 32K or 128K RAM
Analog I/O: daughter card available
Features: DT-connect interface, mezzanine bus for daughter cards
Name: ADSP-21020 System Board
Type: ADSP-21020 IBM PC AT board
Company: Loughborough Sound Images
Processor: ADSP-21020, 160 Kw program RAM, 160 Kw data RAM
Analog I/O: dual 16 bit analog I/O daughter card option.
Features: Interval timer.
_________________________________________________________________
IBM PC boards, AT&T processors:
----------------------------------------------------------------------------
Name: DSP-32C
Type: AT&T DSP-32C IBM PC-AT card
Company: Ariel Corp (908) 249-2900
Processor: DSP-32C
Name: V3-B0-00
Type: AT&T DSP-32C IBM PC-AT (16 bit) card
Company: Communication Automation and Control, Inc. (800) 367-6735
Processor: DSP-32C 50 MHz, 128 K? SRAM (expandable)
Analog I/O: available via mezzanine card
Features: mezzanine bus
Price: $1245
Name: AC5-A0
Type: AT&T DSP-32C IBM PC-AT (16 bit) card
Company: Communication Automation and Control, Inc. (800) 367-6735
Processor: DSP-32C 50 MHz, 64 K? SRAM (expandable)
Analog I/O: available via mezzanine card
Features: mezzanine bus
Price: $995
Name: XC5-A0
Type: AT&T DSP-32C IBM PC-AT (8 bit) card
Company: Communication Automation and Control, Inc. (800) 367-6735
Processor: DSP-32C 49.152 MHz, 64 K? SRAM (expandable)
Analog I/O: available via mezzanine card
Features: mezzanine bus
Price: $995
Name: QUANTUMdsp
Type: DSP3210 IBM PC ISA card
Company: Communication Automation and Control, Inc. (800) 367-6735
Processor: DSP3210 55 MHz, 2 Mbytes DRAM
Analog I/O: 16-bit stereo codec (8 to 48 kHz), telephone line interface
Features: Supports AT&T VCOS DSP operating system, modem/fax/telephony
Name: DSP32C System board
Type: DSP32C IBM PC AT card
Company: Loughborough Sound Images
Processor: DSP32C 50 MHz, 40K words RAM
Analog I/O: dual 16 bit 153 kHz A/D, D/A (Burr Brown PCM78 & PCM56)
Comments: "DSP32 Processor board" is as above, but without analog I/O.
Instead it has a wire wrap prototyping area.
Name: DSP32C Telephony Board
Type: DSP32C IBM PC AT card
Company: Loughborough Sound Images
Processor: DSP32C 50 MHz, 40K words RAM
Analog I/O: AT&T 7525 CODEC
Features: 4 line telephone interface with opto-isolated ring detector in
either US or UK configuration.
Name: AT-DSP2200
Type: AT&T DSP-32C IBM PC-AT card
Company: National Instruments (800) 433-3488
Processor: DSP-32C
Analog I/O: 16 bit sigma-delta A/D, D/A
Features: 16 bit real-time system integration (RTSI) peripheral bus interface.
Name: Array Processor Card AP2
Type: AT&T DSP-32C IBM PC-AT card
Company: Tucker-Davis Technologies
Processor: DSP-32C 50 MHz, 512 Kbytes SRAM, 8.5 Mbytes DRAM
Analog I/O: available via expansion card
Features: fiber optic interface to a variety of peripherals
Comments: available with the "AP2 operating system" (APOS), which is also
a sort of high-level language for rapidly building DSP applications.
Name: Qw3210-SA
Type: DSP3210 IBM PC (ISA) card
Company: Quantawave, (508) 481-9802
Processor: DSP3210 64 MHz, 136 kbytes SRAM, dual-port DRAM between PC, DSP
Analog I/O: dual 16 bit 200 kHz A/D, D/A with programmable gain, cutoff
Price: $2995
Name: MP3210
Type: DSP3210 IBM PC card
Company: Ariel, (908) 249-2900
Processor: 1 or 2 DSP3210 55 MHz, 64 kbytes SRAM (1 ws), 1 Mbyte DRAM
(4-6 ws)
Analog I/O: dual 16 bit 100 kHz A/D, D/A (400 kHz -> 12 bit)
Features: DT-connect interface, NABus interface
Processor: AT&T DSP32C
Host: IBM PC AT (ISA)
RAM: 2 x 32kx32 bits (256 kBytes total)
Analog IO: 2 x 16-bit AD-DA (Sigma-Delta) at +/- 3.0 Volts.
Sampling Rates: 1-50 kHz
Cost: $2250 for educational users. Discounts for volume. Industrial
users by negotiation.
_________________________________________________________________
IBM PC boards, Motorola DSP56000 processors:
----------------------------------------------------------------------------
Name: DSP-56
Type: DSP56001 IBM PC AT card
Company: Ariel Corp (908) 249-2900
Processor: 27 (?) MHz DSP56001
Analog I/O: dual 16 bit 100 kHz A/D, D/A
Features: DSPnet parallel interface, SCSI interface, async serial interface
Name: PC-56
Type: DSP56001 IBM PC AT card
Company: Ariel Corp (908) 249-2900
Processor: 27 MHz 56k with 16K or 64K words RAM
Analog I/O: optional 14 bit single channel analog I/O.
Name: Cheetah mother board
Type: DSP96002/dual DSP56001 IBM PC AT card
Company: Atlanta Signal Processors, Inc. (ASPI) (404) 892-7265
Processor: DSP96002 33 MHz, 256k-2M bytes RAM, dual DSP56001s 20 MHz
Features: serial interfaces.
Name: DSP56001 System board
Type: DSP56001 IBM PC card
Company: Loughborough Sound Images
Processor: DSP56001 20Mhz 192K words RAM
Analog I/O: two 16 bit 153 kHz A/D w/ 3rd order analog filters,
two 16 bit D/A w/ 3rd order analog filters, telephony codec
Comments: "DSP56001 Processor board" is same, but without analog I/O
Name: Dual DSP56001 Processor board
Type: two processor DSP56001 IBM PC card
Company: Loughborough Sound Images
Processor: 27 MHz 56k, 32K words RAM per CPU
Interface: Two ADS ports, each processor has RS422 or SLD ISDN
Comments: 2Kx16 bit dual port RAM between host and each processor,
2Kx24 bit dual port RAM between 56ks
Name: Eagle-56
Type: DSP56002 IBM PC card
Company: Momentum Data Systems
Processor: 66 MHz DSP56002, 256K words RAM
Analog I/O: four 16-bit A/D and D/A pairs, sample rate 5.5 to 48 kHz.
Name: DSP56001 Application Development System (ADS)
Type: DSP56001 board for IBM PC
Company: Motorola Corp. (512) 891-2030
Processor: DSP56001
Comments: Motorola produce DSP56000 assembler, linker, simulator, C compiler.
This is an external board that interfaces to a variety of hosts, including
the IBM PC, via host-specific adaptor cards.
_________________________________________________________________
IBM PC boards, Motorola DSP56156/DSP56116 processors:
----------------------------------------------------------------------------
Name: DSPB56166 System board
Type: DSP56156 IBM PC AT card
Company: Integrated Technologies Solutions, Inc.
Processor: DSP56156 ?? MHz, ?? words RAM,
Features: Xilinx FPGA for custom digital I/O
Name: DSP56116 System board
Type: DSP56116 IBM PC AT card
Company: Spectrum Signal Processing (604) 438-7266
Processor: DSP56116 80? MHz, 30K words RAM,
Analog I/O: two CODECs attached to 56116 serial ports
_________________________________________________________________
IBM PC boards, Motorola DSP96000 processor:
----------------------------------------------------------------------------
Name: MM-96
Type: Two processor DSP96002 IBM PC AT card
Company: Ariel Corp (908) 249-2900
Processor: 2x DSP96002
Name: DSP96002 System board
Type: DSP96002 IBM PC card
Company: Loughborough Sound Images
Processor: DSP96002 33 MHz , 64Kword 0ws RAM + 256Kword 1ws RAM
Analog I/O: dual 16 bit 100 kHz A/D converters (Motorola 56ADC16s)
Name: DSP96002 Application Development System (ADS)
Type: DSP96002 board for IBM PC
Company: Motorola Corp. (512) 891-2030
Processor: DSP96002
Comments: This is an external board that interfaces to a variety of hosts,
including the IBM PC, via host-specific adaptor cards.
_________________________________________________________________
IBM PC boards, Texas Instruments TMS320C1x processors:
----------------------------------------------------------------------------
Name: TMS320C1x Development System
Type: IBM PC card
Company: Loughborough Sound Images
Processor: TMS320C14
Comments: Integrated emulation and programming system.
Type: PC board with TMS32010
Company: Natural Microsystems Corp. (800) 533-6120
Processor: TMS32010
_________________________________________________________________
IBM PC boards, Texas Instruments TMS320C25 processors:
----------------------------------------------------------------------------
Name: DSP-16+
Type: TMS320C25 IBM PC AT card
Company: Ariel Corp (908) 249-2900
Processor: 40 MHz 320C25
Analog I/O: two channel 16 bit 50 kHz A/D, D/A.
Name: Chimera system
Type: TMS320C25 or TMS320C26 IBM PC AT card
Company: Atlanta Signal Processors, Inc. (ASPI) (404) 892-7265
Processor: TMS320C25/TMS320C26, 40/50 MHz, 4K-64K words RAM
Analog I/O: optional 16 bit 200 kHz A/D, D/A is optional via daughter card.
Name: BN2500 DSP development and acquisition processor
Company: Bridgenorth Signal Processing, (604) 538-0003
Type: TMS320C25 IBM PC card
Processor: TMS320C25 40 (opt 50) MHz, 32 Kw 0 ws SRAM, 32Kw 1 ws SRAM,
64 Kw 1 ws SRAM, 256 Kw 1 ws DRAM
Analog I/O: dual 16 bit A/D, D/A via BN3216 expansion module
Features: external buffer management unit to expand address range of C25
Name: Model 250 DSP board
Type: TMS320C25 board for IBM PC AT
Company: Dalanco Spry
Processor: TMS320C25 40 MHz, 4K words program RAM (64K optional),
32K words data RAM (128K optional) (1 ws),
Analog I/O: 12 bit 300 kHz A/D, dual 12 bit 250 kHz D/A
Features: 16 bit expansion bus, serial port
Name: TMS320C25 System board
Type: TMS320C25 IBM PC card
Company: Loughborough Sound Images
Processor: TMS320C25 40 or 50 MHz 16K words RAM
Analog I/O: 1 channel 16 bit 54 kHZ A/D, D/A (100 kHz 12 bit)
Comments: "TMS320C25 Processor board" is as above, but without analog I/O
Name: TMS320C25 Data Acquisition Processor
Type: TMS320C25 IBM PC card
Company: Loughborough Sound Images
Processor: TMS320C25 40 MHz, 64K words program RAM, 64K words data RAM,
256K words buffer RAM
Features: Has buffer management unit, handling circular buffer management,
multiple data channel support, data compare functions, sample counter.
_________________________________________________________________
IBM PC boards, Texas Instruments TMS320C30 processors:
----------------------------------------------------------------------------
Name: Maestro 2110S and 2100D Development Boards
Type: TMS320C30 Development Boards for IBM PC/AT ISA Slot
Company: Wintriss Engineering Corp. (USA), (619) 550-7300 OR (800) 550-7301
weco@powergrid.electriciti.com OR weco@weco.com
Features: Up to 8 MB of Zero Wait-State SRAM
Analog I/O: Two, 96-pin Daughter Card Connectors for Custom Hardware Prototypin
g
Name: MDS-30
Type: PC/AT TMS3203x development board
Company: Traquair Data Systems, Inc.
Processor: TMS320C30 ?? MHz
Name: Banshee mother board
Type: TMS320C30 IBM PC AT card
Company: Atlanta Signal Processors, Inc. (ASPI) (404) 892-7265
Processor: TMS320C30 33 MHz, 256k-512k bytes RAM
Analog I/O: 16 bit 200 kHZ A/D, D/A available via optional boards.
Comments: Optional boards allow multiple Banshee boards to be connected,
provide extra memory, and provide analog I/O.
Name: DSProto Development System
Type: TMS320C31 IBM PC Card
Company: Dicon Lab Inc.
Processor: TMS320C31 33/40/50Mhz 128-768Kbyte SRAM
Analog I/O: 2x16 Bit A/D-D/A 44.1/48 KHz
Comments: serial port/external buffered bus, assembler/linker/loader,
MS Windows debugger
Name: P360F Power Grabber
Type: TMS320C30 board for IBM PC AT
Company: DIPIX Technologies, Inc.
Processor: TMS320C30 33 MHz, 4MB memory, up to 64Mbytes
Analog I/O: RS170/330 and CCIR interlaced, non-int. video inputs,
4 channels, optional display board with TMS 34020 GSP
Features: Digital 16-bit input, DT-Connect interface, selectable pixel clock
Name: BN3000 DSP development and acquisition processor
Company: Bridgenorth Signal Processing, (604) 538-0003
Type: TMS320C30 IBM PC card
Processor: TMS320C30 ?? MHz (60 ns cycle), 512 Kw SRAM (1 ws)
Analog I/O: dual 16 bit A/D, D/A via BN3216 expansion module
Features: serial, parallel interfaces
Name: Tiger 30
Type: TMS320C30 board for IBM PC
Company: DSP Research
Processor: TMS320C30
Name: TMS320C30 System board
Type: TMS320C30 IBM PC AT card
Company: Loughborough Sound Images
Processor: TMS320C30 33 MHz, 64K words RAM
Analog I/O: 2x 16 bit A/D (Burr Brown PCM78), 2x 16 bit D/A (Burr Brown PCM56)
Comments: "TMS320C30 Processor board" is as above, but without analog I/O and
has instead a prototyping area.
Name: SPIRIT-30 AT/ISA
Company: Sonitech International
Type: TMS320C30 IBM PC card
Processor: TMS320C30 33 MHz, 256 Kbytes SRAM (expandable)
Analog I/O: available via serial port interfaces
Features: serial, parallel ports
Name: EVM30
Company: Texas Instruments
Type: TMS320C30 IBM PC card (?)
Processor: TMS320C30 30 MHz, 16 Kw RAM
Analog I/O: 14 bit codec, up to 19.2 kHz sample rate
Comments: inexpensive TI development platform for the C30
_________________________________________________________________
IBM PC boards, Texas Instruments TMS320C40 processors:
----------------------------------------------------------------------------
Name: Vortex system board
Type: TMS320C40/TMS320C31 IBM PC AT card
Company: Atlanta Signal Processors, Inc. (ASPI) (404) 892-7265
Processor: TMS320C40 ?? MHz, TMS320C31 33 MHz, 128 kbytes SRAM
Analog I/O: optional 16 bit 200 kHz A/D, D/A via daughter card.
Name: HEPC2-U
Type: TMS320C40 IBM PC AT card
Company: Traquair Data Systems
Processor: 1 TMS320C40 plus up to 3 TMS320C40 TIM modules
Features: embedded XDS-510 capability to allow connection to external
target
Name: DT3801
Type: TMS320C40 IBM PC AT card
Company: Data Translation
Processor: TMS320C40 40 MHz, 130 Kw SRAM, 4 Mbytes DRAM
Analog I/O: A/D: 12 or 16 bits, 160 kHz to 1 MHz; D/A: 2x 16 bits, 100 kHz
Features: 20 lines digital I/O, 2 counters/timers, separate clocks for A/D, D/A
Name: TMS320C40 Parallel DSP System
Type: IBM PC/AT card
Company: Loughborough Sound Images
Processor: 2 TMS320C40 50 Mhz 64K words RAM
Analog I/O: Daughtercard expansion socketing.
Comments: Uses TIM-40 compatible modules
Name: SPIRIT-40 AT/ISA
Type: TMS320C40 IBM PC AT card
Company: Sonitech International
Processor: TMS320C40 40 MHz, 1 Mbyte SRAM (expandable)
Analog I/O: available via serial port interface
Features: serial, parallel ports
Name: TDB416-40
Type: TMS320C40 PC/AT Development Board
Company: Transtech Parallel Systems Corporation
Processor: TMS320C40 with up to 64 MBytes memory
Features: Also available bundled with 3L Parallel C
Name: TDMB408-40
Type: TMS320C40 PC/AT TIM Motherboard
Company: Transtech Parallel Systems Corporation
Processor: TMS320C40 with up to 8 MBytes memory
and 3 x TMS320C40 TIM-40 module sites
Name: TDMB409
Type: PC/AT TIM Motherboard
Company: Transtech Parallel Systems Corporation
Processor: PC plug-in board with 3 x TMS320C40 TIM-40 module sites
_________________________________________________________________
IBM PC boards, Texas Instruments TMS320C50 processors:
----------------------------------------------------------------------------
Name: TMS320C50 System board
Type: TMS320C50 IBM PC AT card
Company: Loughborough Sound Images
Processor: TMS320C50 50 MHz 32K words RAM
Features: 65sq cm wire wrap area.
_________________________________________________________________
IBM PC boards, Texas Instruments TMS320M500 (Mwave) processor:
----------------------------------------------------------------------------
Name: Mwave multimedia platform
Type: TMS320M500 (Mwave) IBM PC AT card
Company: Atlanta Signal Processors, Inc. (ASPI) (404) 892-7265
Processor: TMS320M500 33(?) MHz, 64 kbytes data SRAM, 96 kbytes prog SRAM
Analog I/O: stereo 16 bit A/D D/A, codec, telephone line interface
Features: MIDI interface, microphone input, speaker output
_________________________________________________________________
IBM PC boards, other processors:
----------------------------------------------------------------------------
Name: PDSP16488 Imaging/Graphics board
Type: PDSP-16488 IBM PC card
Company: Spectrum Signal Processing (604) 438-7266
Processor: PDSP-16488
_________________________________________________________________
Q4.1.2: Mac Nubus DSP development boards.
Mac Nubus boards, AT&T processors:
----------------------------------------------------------------------------
Name: MacDSP MB/A
Type: DSP32C Apple Mac NuBus board
Company: Spectral Innovations (408) 727-1314
Price: $3495-$4995
Processor: DSP32C 60 MHz DSP with up to 1 Mbyte SRAM
Comments: Optional 12 bit A/D card at 16 bit A/D, D/A with 1,2 or 8 channels
_________________________________________________________________
Mac Nubus boards, Motorola DSP56000 processor:
----------------------------------------------------------------------------
Name: MAC-56
Type: Macintosh NuBus DSP56001 card
Company: Ariel Corp. (908) 249-2900
Processor: 56001 with 48-192K words RAM
Comments: Comes with DSPworks software
Name: AudioMedia II
Type: Macintosh NuBus DSP56001 card
Company: DigiDesign Inc. (415) 688-0600
Processor: 56001 @ 33.87 MHz with 8/4/4 K words P/X/Y RAM
Analog I/O: stereo 16 bit 44.1/48 kHz A/D, D/A (?)
Features: SPDIF I/O. Supported by DSP Designer software from Zola
Technologies Inc.
Name: Sound Accelerator II
Type: Macintosh NuBus DSP56001 card
Company: DigiDesign Inc. (415) 688-0600
Processor: 56001 @ 33.87 MHz with 8/16/16 K words P/X/Y RAM
Features: Optional digital/analogue audio I/O unit available. Supported by
DSP Designer software from Zola Technologies Inc.
Name: Pro Tools
Type: Macintosh NuBus 2 x DSP56001 card
Company: DigiDesign Inc. (415) 688-0600
Processor: 2 x 56001 32 MHz, 8/16/16 K words P/X/Y RAM each
Features: Optional digital/analogue audio I/O unit available.
Supported by DSP Designer software from Zola Technologies Inc.
Name of Board: DSP56001 Application Development System (ADS)
Type: DSP56001 board for Mac II
Company: Motorola Corp. (512) 891-2030
Processor: DSP56001
Comments: This is an external board that interfaces to a variety of hosts,
including the Mac, via host-specific adaptor cards.
_________________________________________________________________
Mac Nubus boards, Motorola DSP96000 processor:
----------------------------------------------------------------------------
Name: DSP96002 Application Development System (ADS)
Type: DSP96002 board for Mac II
Company: Motorola Corp. (512) 891-2030
Processor: DSP96002
Comments: This is an external board that interfaces to a variety of hosts,
including the Mac, via host-specific adaptor cards.
_________________________________________________________________
Mac Nubus boards, Texas Instruments processors:
----------------------------------------------------------------------------
Name: NB-DSP2300, 2301, 2305
Type: TMS320C30 board for Mac NuBus
Company: National Instruments (800) 433-3488
Processor: TMS32C30 at 33 (2300), 27 (2301), or 40 (2305) MHz.,
64K or 320K of memory, TMS320C30 to IEEE 754 translator hardware.
Features: 16 bit "real time system integration" (RTSI) peripheral bus.
_________________________________________________________________
Q4.1.3: SBus DSP development boards.
SBus boards, AT&T processors:
----------------------------------------------------------------------------
Name: S-32C
Type: SBus Accelerator Board
Company: Ariel
Price: $2,995 ($5,795 with C compiler)
Processor: AT&T DSP-32C 50MHz, 256 Kbytes to 4 Mbytes SRAM
Analog I/O: Interfaces with Ariel's ProPort & Digital Microphone
Name: MAXCOM Multimedia Processor
Type: Dual DSP3210 Sbus card
Company: KINETICSYSTEMS
Processor: Dual DSP3210
Analog I/O: multimedia audio codec and a telephone quality codec.
Comments: Complete ATT DSP board using real-time operating system (VCOS) and
ATT Multimedia Module Library to provide functions for V.29/V.32 data/fax
modems, speech recognition/synthesis, etc.
_________________________________________________________________
SBus boards, Motorola DSP56000 processor:
----------------------------------------------------------------------------
Name: S-56X
Type: SBus Accelerator Board
Company: Ariel (designed (and also sold) by Berkeley Camera Engineering)
Price: $2,495 / $2,995
Processor: Motorola DSP56001 27 or 32 MHz, 32 Kw RAM
Analog I/O: Digital Microphone, ProPort or AES/EBU
Features: DMA controller, Xilinx 3042 gate array for special purpose I/O.
(1.2 Mby/sec transfer rate using the S-56X, w/ ~3ms latency at DMA page
boundaries).
Name: VS-30
Type: SBus Dual Channel Telephone Signal Processor
Company: Vigra (619) 483-1197
Price: $1375
Availability: In beta test Sep/Nov 1992
Processor: DSP56001 (27 MHz)
Analog I/O: 3 x 8KHz u-Law CODECs. (Line 1, Line 2, handset)
Features: Two RJ-11 telephone jacks, and local handset jack, DTMF transmit
& detect, hook switch control, ring detect, Call progress monitoring
SunOS drivers, programming library, demos. All source code included.
_________________________________________________________________
SBus boards, Texas Instruments TMS320 processors:
----------------------------------------------------------------------------
Name: SPIRIT-30 SBus
Type: 33 MFLOP Application Accelerator -- SBus
Company: Sonitech International Inc. (617) 235-6824
Price: $8,995 ($9,795 for the 40 MFLOP board)
Processor: TMS320C30
Analog I/O: Optional serial port box (extra $1795), IC-100-1 16 bit single
channel @ 100kHz or, DSP/AIB-1: A/D 2 ch., D/A 1 ch., sampling
rate up to 20 kHz,anti-aliasing filters, telephone handset interface
($480). Stereo audio and telephone interface card (1-22) for $695
comes with SW library and demo programs.
Features: Other options for the A/D/A system are available. Filter modules
are extra. Board runs the SPOX operating system. All of their boards are
code compatible with each other (PC and SUNs and VME). Up to six SPIRIT-30
cards can be configured for a 240 MFlop parallel processing system.
Name: SDSP/C30D
Type: TMS320C30 SBus Processor Card
Company: Loughborough Sound Images
Processor: TMS320C30 33 MHz 128Kwords RAM
Comments: also available as SDSP/C30S with SCSI interface
Name: TDMB420
Type: TMS320C40 Sun Sbus Board
Company: Transtech Parallel Systems Corporation
Processor: Sun Sbus to TMS320C40 interface board
Name: HESB40
Type: SBus interface for TMS320C40 TIM-40
Company: Traquair Data Systems, Inc.
Processor: supports one TIM-40 TMS320C40 processors and other external dsps.
Features: Compatible with and capable of controlling HEPC2 PC/AT
board and HEV40 VME board and generic building block.
_________________________________________________________________
SBus boards, other processors:
----------------------------------------------------------------------------
Name: ADDA1418-166
Type: S-Bus A/D and D/A (A/D and D/A are also available separately)
Company: Analyx Systems Inc. (510) 656-8017
Price: $2,495.00 (4-92) (A/D only $1,995, D/A only $1,895)
Processor: None
Analog I/O: 166 kHz 16 channel, 14 bit isolated A/D, 4 Channel 18 bit D/A
Features: 32K Dual-Ported on-board RAM. Single S-Bus slot. All software
included.
_________________________________________________________________
Q4.1.4: VMEbus DSP boards.
VMEbus boards, Analog Devices processors:
----------------------------------------------------------------------------
Name: ADSP-21020 IXD7232
Type: Dual ADSP-21020 VME bus board
Company: Ixthos, Inc.
Processor: 2x ADSP-21020 512 Kbytes data/proc, 196 Kbytes prog/prc
Analog I/O: available via daughter card
Features: I/O mezzanine, RS-232 port, digital I/O option
_________________________________________________________________
VMEbus boards, AT&T processors:
----------------------------------------------------------------------------
Name: SURFboard
Type: VMEbus DSP board
Company: AT&T (Louis Rosa 908-582-5667 or Jim Snyder surf@research.att.com)
Price: $10K
Processor: 6 DSP-32C's organized as 2 sets of 3 processors
(input, output, central -- can be configured)
Analog I/O: None, but can use Ariel ProPort VME board.
Comments: Multiple boards can be chained together. Central DSP's both have
access to a 1Mbyte DRAM which is also the interface to the VMEbus. Input
and output DSP's only have 2Kx32 memory. Central DSP's also have 256Kbytes
of local SRAM.
Name: VME6U6
Type: VMEbus 6U DSP32C board
Company: Communication Automation and Control, Inc. (CACI) (800) 367-6735
Processor: 6 DSP-32C 50 MHz, 512 Kbytes SRAM each
Analog I/O: available via mezzanine bus
Features: TDM expansion port to other boards, mezzanine bus
Price: $10,900
Name: VME9U12
Type: VMEbus 9U DSP32C board
Company: Communication Automation and Control, Inc. (CACI) (800) 367-6735
Processor: 12 DSP-32C 50 MHz, 128 Kbytes SRAM each (exp. to 512 Kby each)
Analog I/O: available via local mezzanine bus
Features: TDM expansion port to other boards, mezzanine bus
Price: $18,800
Name: VE-32C
Type: VMEbus 6U DSP32C four processor board
Company: Valley Technologies
Processor: 4x DSP-32C 50 MHz, 8 Kw SRAM each (exp to 128 Kw each)
Features: 16 bit digital matrix switch for IPC
Name: UltraFFT
Type: VME board (6U form factor)
Company: Valley Technologies
Processor: 2x DSP32C; 2x Plessey PDSP16510 FFT processors
Name: UltraDSP
Type: VME board (6U form factor)
Company: Valley Technologies
Processor: DSP32C 40 MHz; Sharp LH9124 DSP
_________________________________________________________________
VMEbus boards, Motorola processors:
----------------------------------------------------------------------------
Name: V-56 DSP56001 VME board
Type: DSP56001 VME bus card
Company: Ariel Corp. (908) 249-2900
Processor: 27 MHz Motorola DSP56001, 32Kw P RAM, 64Kw data RAM
Features: EPROM boot, 24 bit parallel I/O, serial drivers, 24 bit ADBus
peripheral expansion port.
Analog I/O: None.
Name: DSP 8150
Type: DSP56001 VME bus card
Company: Creative Electronic Systems. Tel: +41 22 7925745. Fax:+41 22 7925748
Email: ces@lancy.ces.ch
Processor: 100 ns Motorola DSP56001, 24-96Kw data RAM
Features: EPROM boot, programmable sequencer, acquisition FIFO, serial
I/O, A24:D32 VME bus access.
Analog I/O: none on board, but connections to a daughter board ADC are provided